From 324752632a2017cc2e2464d110445328ad2a987c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Nov 2013 01:06:15 -0800 Subject: ASoC: rcar: rename GEN2_SRU to GEN2_SCU Gen2 has SCU. SRU is for Gen1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 12afab1..a818ff7 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -18,7 +18,7 @@ #define RSND_GEN1_ADG 1 #define RSND_GEN1_SSI 2 -#define RSND_GEN2_SRU 0 +#define RSND_GEN2_SCU 0 #define RSND_GEN2_ADG 1 #define RSND_GEN2_SSIU 2 #define RSND_GEN2_SSI 3 -- cgit v0.10.2 From e64001e8efc107992fd835770f6383d0dc731594 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 20 Nov 2013 13:17:07 +0000 Subject: ASoC: wm5110: Add extra AIF2 channels Signed-off-by: D.J. Barrow Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 4706d3d..8f4c9d7 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -511,6 +511,38 @@ #define ARIZONA_AIF2TX2MIX_INPUT_3_VOLUME 0x74D #define ARIZONA_AIF2TX2MIX_INPUT_4_SOURCE 0x74E #define ARIZONA_AIF2TX2MIX_INPUT_4_VOLUME 0x74F +#define ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE 0x750 +#define ARIZONA_AIF2TX3MIX_INPUT_1_VOLUME 0x751 +#define ARIZONA_AIF2TX3MIX_INPUT_2_SOURCE 0x752 +#define ARIZONA_AIF2TX3MIX_INPUT_2_VOLUME 0x753 +#define ARIZONA_AIF2TX3MIX_INPUT_3_SOURCE 0x754 +#define ARIZONA_AIF2TX3MIX_INPUT_3_VOLUME 0x755 +#define ARIZONA_AIF2TX3MIX_INPUT_4_SOURCE 0x756 +#define ARIZONA_AIF2TX3MIX_INPUT_4_VOLUME 0x757 +#define ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE 0x758 +#define ARIZONA_AIF2TX4MIX_INPUT_1_VOLUME 0x759 +#define ARIZONA_AIF2TX4MIX_INPUT_2_SOURCE 0x75A +#define ARIZONA_AIF2TX4MIX_INPUT_2_VOLUME 0x75B +#define ARIZONA_AIF2TX4MIX_INPUT_3_SOURCE 0x75C +#define ARIZONA_AIF2TX4MIX_INPUT_3_VOLUME 0x75D +#define ARIZONA_AIF2TX4MIX_INPUT_4_SOURCE 0x75E +#define ARIZONA_AIF2TX4MIX_INPUT_4_VOLUME 0x75F +#define ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE 0x760 +#define ARIZONA_AIF2TX5MIX_INPUT_1_VOLUME 0x761 +#define ARIZONA_AIF2TX5MIX_INPUT_2_SOURCE 0x762 +#define ARIZONA_AIF2TX5MIX_INPUT_2_VOLUME 0x763 +#define ARIZONA_AIF2TX5MIX_INPUT_3_SOURCE 0x764 +#define ARIZONA_AIF2TX5MIX_INPUT_3_VOLUME 0x765 +#define ARIZONA_AIF2TX5MIX_INPUT_4_SOURCE 0x766 +#define ARIZONA_AIF2TX5MIX_INPUT_4_VOLUME 0x767 +#define ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE 0x768 +#define ARIZONA_AIF2TX6MIX_INPUT_1_VOLUME 0x769 +#define ARIZONA_AIF2TX6MIX_INPUT_2_SOURCE 0x76A +#define ARIZONA_AIF2TX6MIX_INPUT_2_VOLUME 0x76B +#define ARIZONA_AIF2TX6MIX_INPUT_3_SOURCE 0x76C +#define ARIZONA_AIF2TX6MIX_INPUT_3_VOLUME 0x76D +#define ARIZONA_AIF2TX6MIX_INPUT_4_SOURCE 0x76E +#define ARIZONA_AIF2TX6MIX_INPUT_4_VOLUME 0x76F #define ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE 0x780 #define ARIZONA_AIF3TX1MIX_INPUT_1_VOLUME 0x781 #define ARIZONA_AIF3TX1MIX_INPUT_2_SOURCE 0x782 @@ -3726,6 +3758,35 @@ #define ARIZONA_AIF2TX2_SLOT_WIDTH 6 /* AIF2TX2_SLOT - [5:0] */ /* + * R1355 (0x54B) - AIF2 Frame Ctrl 5 + */ +#define ARIZONA_AIF2TX3_SLOT_MASK 0x003F /* AIF2TX3_SLOT - [5:0] */ +#define ARIZONA_AIF2TX3_SLOT_SHIFT 0 /* AIF2TX3_SLOT - [5:0] */ +#define ARIZONA_AIF2TX3_SLOT_WIDTH 6 /* AIF2TX3_SLOT - [5:0] */ + +/* + * R1356 (0x54C) - AIF2 Frame Ctrl 6 + */ +#define ARIZONA_AIF2TX4_SLOT_MASK 0x003F /* AIF2TX4_SLOT - [5:0] */ +#define ARIZONA_AIF2TX4_SLOT_SHIFT 0 /* AIF2TX4_SLOT - [5:0] */ +#define ARIZONA_AIF2TX4_SLOT_WIDTH 6 /* AIF2TX4_SLOT - [5:0] */ + + +/* + * R1357 (0x54D) - AIF2 Frame Ctrl 7 + */ +#define ARIZONA_AIF2TX5_SLOT_MASK 0x003F /* AIF2TX5_SLOT - [5:0] */ +#define ARIZONA_AIF2TX5_SLOT_SHIFT 0 /* AIF2TX5_SLOT - [5:0] */ +#define ARIZONA_AIF2TX5_SLOT_WIDTH 6 /* AIF2TX5_SLOT - [5:0] */ + +/* + * R1358 (0x54E) - AIF2 Frame Ctrl 8 + */ +#define ARIZONA_AIF2TX6_SLOT_MASK 0x003F /* AIF2TX6_SLOT - [5:0] */ +#define ARIZONA_AIF2TX6_SLOT_SHIFT 0 /* AIF2TX6_SLOT - [5:0] */ +#define ARIZONA_AIF2TX6_SLOT_WIDTH 6 /* AIF2TX6_SLOT - [5:0] */ + +/* * R1361 (0x551) - AIF2 Frame Ctrl 11 */ #define ARIZONA_AIF2RX1_SLOT_MASK 0x003F /* AIF2RX1_SLOT - [5:0] */ @@ -3740,8 +3801,52 @@ #define ARIZONA_AIF2RX2_SLOT_WIDTH 6 /* AIF2RX2_SLOT - [5:0] */ /* + * R1363 (0x553) - AIF2 Frame Ctrl 13 + */ +#define ARIZONA_AIF2RX3_SLOT_MASK 0x003F /* AIF2RX3_SLOT - [5:0] */ +#define ARIZONA_AIF2RX3_SLOT_SHIFT 0 /* AIF2RX3_SLOT - [5:0] */ +#define ARIZONA_AIF2RX3_SLOT_WIDTH 6 /* AIF2RX3_SLOT - [5:0] */ + +/* + * R1364 (0x554) - AIF2 Frame Ctrl 14 + */ +#define ARIZONA_AIF2RX4_SLOT_MASK 0x003F /* AIF2RX4_SLOT - [5:0] */ +#define ARIZONA_AIF2RX4_SLOT_SHIFT 0 /* AIF2RX4_SLOT - [5:0] */ +#define ARIZONA_AIF2RX4_SLOT_WIDTH 6 /* AIF2RX4_SLOT - [5:0] */ + +/* + * R1365 (0x555) - AIF2 Frame Ctrl 15 + */ +#define ARIZONA_AIF2RX5_SLOT_MASK 0x003F /* AIF2RX5_SLOT - [5:0] */ +#define ARIZONA_AIF2RX5_SLOT_SHIFT 0 /* AIF2RX5_SLOT - [5:0] */ +#define ARIZONA_AIF2RX5_SLOT_WIDTH 6 /* AIF2RX5_SLOT - [5:0] */ + +/* + * R1366 (0x556) - AIF2 Frame Ctrl 16 + */ +#define ARIZONA_AIF2RX6_SLOT_MASK 0x003F /* AIF2RX6_SLOT - [5:0] */ +#define ARIZONA_AIF2RX6_SLOT_SHIFT 0 /* AIF2RX6_SLOT - [5:0] */ +#define ARIZONA_AIF2RX6_SLOT_WIDTH 6 /* AIF2RX6_SLOT - [5:0] */ + +/* * R1369 (0x559) - AIF2 Tx Enables */ +#define ARIZONA_AIF2TX6_ENA 0x0020 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX6_ENA_MASK 0x0020 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX6_ENA_SHIFT 5 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX6_ENA_WIDTH 1 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX5_ENA 0x0010 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX5_ENA_MASK 0x0010 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX5_ENA_SHIFT 4 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX5_ENA_WIDTH 1 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX4_ENA 0x0008 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX4_ENA_MASK 0x0008 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX4_ENA_SHIFT 3 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX4_ENA_WIDTH 1 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX3_ENA 0x0004 /* AIF2TX3_ENA */ +#define ARIZONA_AIF2TX3_ENA_MASK 0x0004 /* AIF2TX3_ENA */ +#define ARIZONA_AIF2TX3_ENA_SHIFT 2 /* AIF2TX3_ENA */ +#define ARIZONA_AIF2TX3_ENA_WIDTH 1 /* AIF2TX3_ENA */ #define ARIZONA_AIF2TX2_ENA 0x0002 /* AIF2TX2_ENA */ #define ARIZONA_AIF2TX2_ENA_MASK 0x0002 /* AIF2TX2_ENA */ #define ARIZONA_AIF2TX2_ENA_SHIFT 1 /* AIF2TX2_ENA */ @@ -3754,6 +3859,22 @@ /* * R1370 (0x55A) - AIF2 Rx Enables */ +#define ARIZONA_AIF2RX6_ENA 0x0020 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX6_ENA_MASK 0x0020 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX6_ENA_SHIFT 5 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX6_ENA_WIDTH 1 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX5_ENA 0x0010 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX5_ENA_MASK 0x0010 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX5_ENA_SHIFT 4 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX5_ENA_WIDTH 1 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX4_ENA 0x0008 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX4_ENA_MASK 0x0008 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX4_ENA_SHIFT 3 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX4_ENA_WIDTH 1 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX3_ENA 0x0004 /* AIF2RX3_ENA */ +#define ARIZONA_AIF2RX3_ENA_MASK 0x0004 /* AIF2RX3_ENA */ +#define ARIZONA_AIF2RX3_ENA_SHIFT 2 /* AIF2RX3_ENA */ +#define ARIZONA_AIF2RX3_ENA_WIDTH 1 /* AIF2RX3_ENA */ #define ARIZONA_AIF2RX2_ENA 0x0002 /* AIF2RX2_ENA */ #define ARIZONA_AIF2RX2_ENA_MASK 0x0002 /* AIF2RX2_ENA */ #define ARIZONA_AIF2RX2_ENA_SHIFT 1 /* AIF2RX2_ENA */ diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6f05b17..6977bf9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -292,6 +292,10 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "AIF1RX8", "AIF2RX1", "AIF2RX2", + "AIF2RX3", + "AIF2RX4", + "AIF2RX5", + "AIF2RX6", "AIF3RX1", "AIF3RX2", "SLIMRX1", @@ -395,6 +399,10 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x27, 0x28, /* AIF2RX1 */ 0x29, + 0x2a, + 0x2b, + 0x2c, + 0x2d, 0x30, /* AIF3RX1 */ 0x31, 0x38, /* SLIMRX1 */ diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9e81b63..1f96672 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -81,7 +81,7 @@ struct arizona_priv { unsigned int spk_ena_pending:1; }; -#define ARIZONA_NUM_MIXER_INPUTS 99 +#define ARIZONA_NUM_MIXER_INPUTS 103 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd6438..181de7d 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -302,6 +302,10 @@ ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), @@ -361,6 +365,10 @@ ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); @@ -561,11 +569,27 @@ SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, @@ -703,6 +727,10 @@ ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), +ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"), +ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), @@ -764,6 +792,10 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF1RX8", "AIF1RX8" }, \ { name, "AIF2RX1", "AIF2RX1" }, \ { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF2RX5", "AIF2RX5" }, \ + { name, "AIF2RX6", "AIF2RX6" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ { name, "SLIMRX1", "SLIMRX1" }, \ @@ -861,9 +893,17 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "AIF2TX1" }, { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + { "AIF2 Capture", NULL, "AIF2TX5" }, + { "AIF2 Capture", NULL, "AIF2TX6" }, { "AIF2RX1", NULL, "AIF2 Playback" }, { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + { "AIF2RX5", NULL, "AIF2 Playback" }, + { "AIF2RX6", NULL, "AIF2 Playback" }, { "AIF3 Capture", NULL, "AIF3TX1" }, { "AIF3 Capture", NULL, "AIF3TX2" }, @@ -947,6 +987,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"), + ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"), ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), @@ -1067,14 +1111,14 @@ static struct snd_soc_dai_driver wm5110_dai[] = { .playback = { .stream_name = "AIF2 Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 6, .rates = WM5110_RATES, .formats = WM5110_FORMATS, }, .capture = { .stream_name = "AIF2 Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 6, .rates = WM5110_RATES, .formats = WM5110_FORMATS, }, -- cgit v0.10.2 From c50aa44db3f803d3c3b79c926e76862454e77c48 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Nov 2013 16:04:00 +0000 Subject: mfd: wm5110: Make DSP memories readable Expose the memory regions used by the DSP cores on WM5110 as readable and volatile. Signed-off-by: Charles Keepax Signed-off-by: Lee Jones diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 3113e39..4430404 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -14,6 +14,7 @@ #include #include +#include #include "arizona.h" @@ -1336,6 +1337,64 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00001404, 0x0000 }, /* R5124 - DSP4 Status 1 */ }; +static bool wm5110_is_rev_b_adsp_memory(unsigned int reg) +{ + if ((reg >= 0x100000 && reg < 0x103000) || + (reg >= 0x180000 && reg < 0x181000) || + (reg >= 0x190000 && reg < 0x192000) || + (reg >= 0x1a8000 && reg < 0x1a9000) || + (reg >= 0x200000 && reg < 0x209000) || + (reg >= 0x280000 && reg < 0x281000) || + (reg >= 0x290000 && reg < 0x29a000) || + (reg >= 0x2a8000 && reg < 0x2aa000) || + (reg >= 0x300000 && reg < 0x30f000) || + (reg >= 0x380000 && reg < 0x382000) || + (reg >= 0x390000 && reg < 0x39e000) || + (reg >= 0x3a8000 && reg < 0x3b6000) || + (reg >= 0x400000 && reg < 0x403000) || + (reg >= 0x480000 && reg < 0x481000) || + (reg >= 0x490000 && reg < 0x492000) || + (reg >= 0x4a8000 && reg < 0x4a9000)) + return true; + else + return false; +} + +static bool wm5110_is_rev_d_adsp_memory(unsigned int reg) +{ + if ((reg >= 0x100000 && reg < 0x106000) || + (reg >= 0x180000 && reg < 0x182000) || + (reg >= 0x190000 && reg < 0x198000) || + (reg >= 0x1a8000 && reg < 0x1aa000) || + (reg >= 0x200000 && reg < 0x20f000) || + (reg >= 0x280000 && reg < 0x282000) || + (reg >= 0x290000 && reg < 0x29c000) || + (reg >= 0x2a6000 && reg < 0x2b4000) || + (reg >= 0x300000 && reg < 0x30f000) || + (reg >= 0x380000 && reg < 0x382000) || + (reg >= 0x390000 && reg < 0x3a2000) || + (reg >= 0x3a6000 && reg < 0x3b4000) || + (reg >= 0x400000 && reg < 0x406000) || + (reg >= 0x480000 && reg < 0x482000) || + (reg >= 0x490000 && reg < 0x498000) || + (reg >= 0x4a8000 && reg < 0x4aa000)) + return true; + else + return false; +} + +static bool wm5110_is_adsp_memory(struct device *dev, unsigned int reg) +{ + struct arizona *arizona = dev_get_drvdata(dev); + + switch (arizona->rev) { + case 0 ... 2: + return wm5110_is_rev_b_adsp_memory(reg); + default: + return wm5110_is_rev_d_adsp_memory(reg); + } +} + static bool wm5110_readable_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -2308,7 +2367,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_DSP4_STATUS_3: return true; default: - return false; + return wm5110_is_adsp_memory(dev, reg); } } @@ -2368,16 +2427,18 @@ static bool wm5110_volatile_register(struct device *dev, unsigned int reg) case ARIZONA_DSP4_STATUS_3: return true; default: - return false; + return wm5110_is_adsp_memory(dev, reg); } } +#define WM5110_MAX_REGISTER 0x4a9fff + const struct regmap_config wm5110_spi_regmap = { .reg_bits = 32, .pad_bits = 16, .val_bits = 16, - .max_register = ARIZONA_DSP1_STATUS_2, + .max_register = WM5110_MAX_REGISTER, .readable_reg = wm5110_readable_register, .volatile_reg = wm5110_volatile_register, @@ -2391,7 +2452,7 @@ const struct regmap_config wm5110_i2c_regmap = { .reg_bits = 32, .val_bits = 16, - .max_register = ARIZONA_DSP1_STATUS_2, + .max_register = WM5110_MAX_REGISTER, .readable_reg = wm5110_readable_register, .volatile_reg = wm5110_volatile_register, -- cgit v0.10.2 From 254dc326dbfd23c2678fafad1b84fc0e42ac4374 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Nov 2013 16:04:03 +0000 Subject: ASoC: wm5110: Expose input high pass filter controls Acked-by: Mark Brown Signed-off-by: Charles Keepax Signed-off-by: Lee Jones diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 4430404..3b079f6 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -518,6 +518,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000300, 0x0000 }, /* R768 - Input Enables */ { 0x00000308, 0x0000 }, /* R776 - Input Rate */ { 0x00000309, 0x0022 }, /* R777 - Input Volume Ramp */ + { 0x0000030C, 0x0002 }, /* R780 - HPF Control */ { 0x00000310, 0x2080 }, /* R784 - IN1L Control */ { 0x00000311, 0x0180 }, /* R785 - ADC Digital Volume 1L */ { 0x00000312, 0x0000 }, /* R786 - DMIC1L Control */ @@ -539,6 +540,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000328, 0x2000 }, /* R808 - IN4L Control */ { 0x00000329, 0x0180 }, /* R809 - ADC Digital Volume 4L */ { 0x0000032A, 0x0000 }, /* R810 - DMIC4L Control */ + { 0x0000032C, 0x0000 }, /* R812 - IN4R Control */ { 0x0000032D, 0x0180 }, /* R813 - ADC Digital Volume 4R */ { 0x0000032E, 0x0000 }, /* R814 - DMIC4R Control */ { 0x00000400, 0x0000 }, /* R1024 - Output Enables 1 */ @@ -1512,6 +1514,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_INPUT_ENABLES_STATUS: case ARIZONA_INPUT_RATE: case ARIZONA_INPUT_VOLUME_RAMP: + case ARIZONA_HPF_CONTROL: case ARIZONA_IN1L_CONTROL: case ARIZONA_ADC_DIGITAL_VOLUME_1L: case ARIZONA_DMIC1L_CONTROL: @@ -1533,6 +1536,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_IN4L_CONTROL: case ARIZONA_ADC_DIGITAL_VOLUME_4L: case ARIZONA_DMIC4L_CONTROL: + case ARIZONA_IN4R_CONTROL: case ARIZONA_ADC_DIGITAL_VOLUME_4R: case ARIZONA_DMIC4R_CONTROL: case ARIZONA_OUTPUT_ENABLES_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 4706d3d..cdf1f5ac 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -139,6 +139,7 @@ #define ARIZONA_INPUT_ENABLES_STATUS 0x301 #define ARIZONA_INPUT_RATE 0x308 #define ARIZONA_INPUT_VOLUME_RAMP 0x309 +#define ARIZONA_HPF_CONTROL 0x30C #define ARIZONA_IN1L_CONTROL 0x310 #define ARIZONA_ADC_DIGITAL_VOLUME_1L 0x311 #define ARIZONA_DMIC1L_CONTROL 0x312 @@ -160,6 +161,7 @@ #define ARIZONA_IN4L_CONTROL 0x328 #define ARIZONA_ADC_DIGITAL_VOLUME_4L 0x329 #define ARIZONA_DMIC4L_CONTROL 0x32A +#define ARIZONA_IN4R_CONTROL 0x32C #define ARIZONA_ADC_DIGITAL_VOLUME_4R 0x32D #define ARIZONA_DMIC4R_CONTROL 0x32E #define ARIZONA_OUTPUT_ENABLES_1 0x400 @@ -2293,8 +2295,18 @@ #define ARIZONA_IN_VI_RAMP_WIDTH 3 /* IN_VI_RAMP - [2:0] */ /* + * R780 (0x30C) - HPF Control + */ +#define ARIZONA_IN_HPF_CUT_MASK 0x0007 /* IN_HPF_CUT [2:0] */ +#define ARIZONA_IN_HPF_CUT_SHIFT 0 /* IN_HPF_CUT [2:0] */ +#define ARIZONA_IN_HPF_CUT_WIDTH 3 /* IN_HPF_CUT [2:0] */ + +/* * R784 (0x310) - IN1L Control */ +#define ARIZONA_IN1L_HPF_MASK 0x8000 /* IN1L_HPF - [15] */ +#define ARIZONA_IN1L_HPF_SHIFT 15 /* IN1L_HPF - [15] */ +#define ARIZONA_IN1L_HPF_WIDTH 1 /* IN1L_HPF - [15] */ #define ARIZONA_IN1_OSR_MASK 0x6000 /* IN1_OSR - [14:13] */ #define ARIZONA_IN1_OSR_SHIFT 13 /* IN1_OSR - [14:13] */ #define ARIZONA_IN1_OSR_WIDTH 2 /* IN1_OSR - [14:13] */ @@ -2333,6 +2345,9 @@ /* * R788 (0x314) - IN1R Control */ +#define ARIZONA_IN1R_HPF_MASK 0x8000 /* IN1R_HPF - [15] */ +#define ARIZONA_IN1R_HPF_SHIFT 15 /* IN1R_HPF - [15] */ +#define ARIZONA_IN1R_HPF_WIDTH 1 /* IN1R_HPF - [15] */ #define ARIZONA_IN1R_PGA_VOL_MASK 0x00FE /* IN1R_PGA_VOL - [7:1] */ #define ARIZONA_IN1R_PGA_VOL_SHIFT 1 /* IN1R_PGA_VOL - [7:1] */ #define ARIZONA_IN1R_PGA_VOL_WIDTH 7 /* IN1R_PGA_VOL - [7:1] */ @@ -2362,6 +2377,9 @@ /* * R792 (0x318) - IN2L Control */ +#define ARIZONA_IN2L_HPF_MASK 0x8000 /* IN2L_HPF - [15] */ +#define ARIZONA_IN2L_HPF_SHIFT 15 /* IN2L_HPF - [15] */ +#define ARIZONA_IN2L_HPF_WIDTH 1 /* IN2L_HPF - [15] */ #define ARIZONA_IN2_OSR_MASK 0x6000 /* IN2_OSR - [14:13] */ #define ARIZONA_IN2_OSR_SHIFT 13 /* IN2_OSR - [14:13] */ #define ARIZONA_IN2_OSR_WIDTH 2 /* IN2_OSR - [14:13] */ @@ -2400,6 +2418,9 @@ /* * R796 (0x31C) - IN2R Control */ +#define ARIZONA_IN2R_HPF_MASK 0x8000 /* IN2R_HPF - [15] */ +#define ARIZONA_IN2R_HPF_SHIFT 15 /* IN2R_HPF - [15] */ +#define ARIZONA_IN2R_HPF_WIDTH 1 /* IN2R_HPF - [15] */ #define ARIZONA_IN2R_PGA_VOL_MASK 0x00FE /* IN2R_PGA_VOL - [7:1] */ #define ARIZONA_IN2R_PGA_VOL_SHIFT 1 /* IN2R_PGA_VOL - [7:1] */ #define ARIZONA_IN2R_PGA_VOL_WIDTH 7 /* IN2R_PGA_VOL - [7:1] */ @@ -2429,6 +2450,9 @@ /* * R800 (0x320) - IN3L Control */ +#define ARIZONA_IN3L_HPF_MASK 0x8000 /* IN3L_HPF - [15] */ +#define ARIZONA_IN3L_HPF_SHIFT 15 /* IN3L_HPF - [15] */ +#define ARIZONA_IN3L_HPF_WIDTH 1 /* IN3L_HPF - [15] */ #define ARIZONA_IN3_OSR_MASK 0x6000 /* IN3_OSR - [14:13] */ #define ARIZONA_IN3_OSR_SHIFT 13 /* IN3_OSR - [14:13] */ #define ARIZONA_IN3_OSR_WIDTH 2 /* IN3_OSR - [14:13] */ @@ -2467,6 +2491,9 @@ /* * R804 (0x324) - IN3R Control */ +#define ARIZONA_IN3R_HPF_MASK 0x8000 /* IN3R_HPF - [15] */ +#define ARIZONA_IN3R_HPF_SHIFT 15 /* IN3R_HPF - [15] */ +#define ARIZONA_IN3R_HPF_WIDTH 1 /* IN3R_HPF - [15] */ #define ARIZONA_IN3R_PGA_VOL_MASK 0x00FE /* IN3R_PGA_VOL - [7:1] */ #define ARIZONA_IN3R_PGA_VOL_SHIFT 1 /* IN3R_PGA_VOL - [7:1] */ #define ARIZONA_IN3R_PGA_VOL_WIDTH 7 /* IN3R_PGA_VOL - [7:1] */ @@ -2496,6 +2523,9 @@ /* * R808 (0x328) - IN4 Control */ +#define ARIZONA_IN4L_HPF_MASK 0x8000 /* IN4L_HPF - [15] */ +#define ARIZONA_IN4L_HPF_SHIFT 15 /* IN4L_HPF - [15] */ +#define ARIZONA_IN4L_HPF_WIDTH 1 /* IN4L_HPF - [15] */ #define ARIZONA_IN4_OSR_MASK 0x6000 /* IN4_OSR - [14:13] */ #define ARIZONA_IN4_OSR_SHIFT 13 /* IN4_OSR - [14:13] */ #define ARIZONA_IN4_OSR_WIDTH 2 /* IN4_OSR - [14:13] */ @@ -2526,6 +2556,13 @@ #define ARIZONA_IN4L_DMIC_DLY_WIDTH 6 /* IN4L_DMIC_DLY - [5:0] */ /* + * R812 (0x32C) - IN4R Control + */ +#define ARIZONA_IN4R_HPF_MASK 0x8000 /* IN4R_HPF - [15] */ +#define ARIZONA_IN4R_HPF_SHIFT 15 /* IN4R_HPF - [15] */ +#define ARIZONA_IN4R_HPF_WIDTH 1 /* IN4R_HPF - [15] */ + +/* * R813 (0x32D) - ADC Digital Volume 4R */ #define ARIZONA_IN_VU 0x0200 /* IN_VU */ diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808b..7083262 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -560,6 +560,16 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_hpf_cut_text[] = { + "2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz" +}; + +const struct soc_enum arizona_in_hpf_cut_enum = + SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT, + ARRAY_SIZE(arizona_in_hpf_cut_text), + arizona_in_hpf_cut_text); +EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); + static const char * const arizona_in_dmic_osr_text[] = { "1.536MHz", "3.072MHz", "6.144MHz", }; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9e81b63..f8e6386 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -199,6 +199,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_hpf_cut_enum; extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd6438..ea18e88 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -76,6 +76,25 @@ SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), + +SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2L HPF Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2R HPF Switch", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN3L HPF Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN3R HPF Switch", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN4L HPF Switch", ARIZONA_IN4L_CONTROL, + ARIZONA_IN4L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN4R HPF Switch", ARIZONA_IN4R_CONTROL, + ARIZONA_IN4R_HPF_SHIFT, 1, 0), + SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, -- cgit v0.10.2 From 74c375cb85d7374734e6e53af41c724d9a937f8e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:43 -0200 Subject: ASoC: ad193x: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index aea7e52..12c27eb 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,7 +413,7 @@ static struct spi_driver ad193x_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static const struct regmap_config ad193x_i2c_regmap_config = { .val_bits = 8, @@ -470,7 +470,7 @@ static int __init ad193x_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&ad193x_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n", @@ -495,7 +495,7 @@ static void __exit ad193x_modexit(void) spi_unregister_driver(&ad193x_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&ad193x_i2c_driver); #endif } -- cgit v0.10.2 From 04c3a852f51ff40f32a29e14078432038b5bcdbc Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:44 -0200 Subject: ASoC: adav80x: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 14a7c16..f7bf455 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -939,7 +939,7 @@ static struct spi_driver adav80x_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static const struct regmap_config adav80x_i2c_regmap_config = { .val_bits = 8, .pad_bits = 1, @@ -985,7 +985,7 @@ static int __init adav80x_init(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&adav80x_i2c_driver); if (ret) return ret; @@ -1001,7 +1001,7 @@ module_init(adav80x_init); static void __exit adav80x_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&adav80x_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From b34d7cf355116f5107fad8d42fb91d067a3b15bc Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:46 -0200 Subject: ASoC: ak4642: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 090d499..2f861c9 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -511,7 +511,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -576,7 +576,7 @@ static struct i2c_driver ak4642_i2c_driver = { static int __init ak4642_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&ak4642_i2c_driver); #endif return ret; @@ -586,7 +586,7 @@ module_init(ak4642_modinit); static void __exit ak4642_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&ak4642_i2c_driver); #endif -- cgit v0.10.2 From d8764646e1cc0ad209af29396e59f836b7d8f164 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Wed, 20 Nov 2013 10:04:15 +0100 Subject: ASoC: fsl-ssi: Move ac97 specific setup to seperate function This is a pure cleanup patch to increase code readability. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 35e2773..fb8f52a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -321,6 +321,36 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) return ret; } +static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) +{ + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + /* + * Setup the clock control register + */ + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->stccr); + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->srccr); + + /* + * Enable AC97 mode and startup the SSI + */ + write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV, + &ssi->sacnt); + write_ssi(0xff, &ssi->saccdis); + write_ssi(0x300, &ssi->saccen); + + /* + * Enable SSI, Transmit and Receive. AC97 has to communicate with the + * codec before a stream is started. + */ + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | + CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); + + write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); +} + static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; @@ -387,31 +417,8 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) * because it is also running without an active substream. Normally SSI * is only enabled when there is a substream. */ - if (ssi_private->imx_ac97) { - /* - * Setup the clock control register - */ - write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), - &ssi->stccr); - write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), - &ssi->srccr); - - /* - * Enable AC97 mode and startup the SSI - */ - write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV, - &ssi->sacnt); - write_ssi(0xff, &ssi->saccdis); - write_ssi(0x300, &ssi->saccen); - - /* - * Enable SSI, Transmit and Receive - */ - write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | - CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); - - write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); - } + if (ssi_private->imx_ac97) + fsl_ssi_setup_ac97(ssi_private); return 0; } -- cgit v0.10.2 From c600e95360dac3a3b88f0a2106214dff8e5f56be Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 19 Nov 2013 14:12:25 +0200 Subject: ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format The new playback format is needed for tda998x HDMI audio support. At the moment the only other user of this codec is omap-hdmi-audio. This change should not break anything because omap-hdmi-audio-dai, the cpu-dai of omap-hdmi-audio, enforces sufficient constraints to available sample formats. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 68342b1..32797a8 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -44,7 +44,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { .stream_name = "Capture", -- cgit v0.10.2 From 1b488a481c39d9cd36535b6c15fe474546e6460b Mon Sep 17 00:00:00 2001 From: Victor Kamensky Date: Sat, 16 Nov 2013 02:01:19 +0200 Subject: ASoC: omap: mcbsp, mcpdm, dmic: raw read and write endian fix All OMAP IP blocks expect LE data, but CPU may operate in BE mode. Need to use endian neutral functions to read/write h/w registers. I.e instead of __raw_read[lw] and __raw_write[lw] functions code need to use read[lw]_relaxed and write[lw]_relaxed functions. If the first simply reads/writes register, the second will byteswap it if host operates in BE mode. Changes are trivial sed like replacement of __raw_xxx functions with xxx_relaxed variant. Signed-off-by: Victor Kamensky Signed-off-by: Taras Kondratiuk Acked-by: Jarkko Nikula Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 83433fd..86c7538 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -36,10 +36,10 @@ static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) if (mcbsp->pdata->reg_size == 2) { ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; - __raw_writew((u16)val, addr); + writew_relaxed((u16)val, addr); } else { ((u32 *)mcbsp->reg_cache)[reg] = val; - __raw_writel(val, addr); + writel_relaxed(val, addr); } } @@ -48,22 +48,22 @@ static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache) void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; if (mcbsp->pdata->reg_size == 2) { - return !from_cache ? __raw_readw(addr) : + return !from_cache ? readw_relaxed(addr) : ((u16 *)mcbsp->reg_cache)[reg]; } else { - return !from_cache ? __raw_readl(addr) : + return !from_cache ? readl_relaxed(addr) : ((u32 *)mcbsp->reg_cache)[reg]; } } static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) { - __raw_writel(val, mcbsp->st_data->io_base_st + reg); + writel_relaxed(val, mcbsp->st_data->io_base_st + reg); } static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) { - return __raw_readl(mcbsp->st_data->io_base_st + reg); + return readl_relaxed(mcbsp->st_data->io_base_st + reg); } #define MCBSP_READ(mcbsp, reg) \ diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 12e566b..1bd531d 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -61,12 +61,12 @@ struct omap_dmic { static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) { - __raw_writel(val, dmic->io_base + reg); + writel_relaxed(val, dmic->io_base + reg); } static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg) { - return __raw_readl(dmic->io_base + reg); + return readl_relaxed(dmic->io_base + reg); } static inline void omap_dmic_start(struct omap_dmic *dmic) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index cd9ee16..2f5b153 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -74,12 +74,12 @@ struct omap_mcpdm { static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val) { - __raw_writel(val, mcpdm->io_base + reg); + writel_relaxed(val, mcpdm->io_base + reg); } static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg) { - return __raw_readl(mcpdm->io_base + reg); + return readl_relaxed(mcpdm->io_base + reg); } #ifdef DEBUG -- cgit v0.10.2 From a37377314ff068c83f425979142263a17a6f18af Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 8 Nov 2013 12:46:53 +0300 Subject: ASoC: rcar: some dubious one-bit signed bitfields Because these are signed they can either be 0 or -1 instead of 0 and 1 as intended. It doesn't cause a problem from what I can see, but it's dangerous and Sparse complains: sound/soc/sh/rcar/rsnd.h:177:25: error: dubious one-bit signed bitfield Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9e463e5..b5ac3a2 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -174,11 +174,11 @@ struct rsnd_dai { struct rsnd_dai_stream playback; struct rsnd_dai_stream capture; - int clk_master:1; - int bit_clk_inv:1; - int frm_clk_inv:1; - int sys_delay:1; - int data_alignment:1; + unsigned int clk_master:1; + unsigned int bit_clk_inv:1; + unsigned int frm_clk_inv:1; + unsigned int sys_delay:1; + unsigned int data_alignment:1; }; #define rsnd_dai_nr(priv) ((priv)->dai_nr) -- cgit v0.10.2 From 06b2bd23057f9dad149f0dda436c7426c87b986f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:52 -0200 Subject: ASoC: uda1380: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index fd0a314..726df6d 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -794,7 +794,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .num_dapm_routes = ARRAY_SIZE(uda1380_dapm_routes), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int uda1380_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -840,7 +840,7 @@ static struct i2c_driver uda1380_i2c_driver = { static int __init uda1380_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); @@ -851,7 +851,7 @@ module_init(uda1380_modinit); static void __exit uda1380_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&uda1380_i2c_driver); #endif } -- cgit v0.10.2 From 784cbf8ab4641c874806a938b9c863c91d70fbe5 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 21 Nov 2013 13:32:24 +0200 Subject: ASoC: Rename mid-x86 directory to intel We have other Intel platforms coming having the Smart Sound Technology (SST) so rename the mid-x86 directory to intel as originally directory name reflected only Intel MID platform. Signed-off-by: Jarkko Nikula Acked-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b84..463a9e2 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -42,7 +42,7 @@ source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" -source "sound/soc/mid-x86/Kconfig" +source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e701..ff291d3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -16,7 +16,7 @@ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ -obj-$(CONFIG_SND_SOC) += mid-x86/ +obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig new file mode 100644 index 0000000..61c10bf --- /dev/null +++ b/sound/soc/intel/Kconfig @@ -0,0 +1,13 @@ +config SND_MFLD_MACHINE + tristate "SOC Machine Audio driver for Intel Medfield MID platform" + depends on INTEL_SCU_IPC + select SND_SOC_SN95031 + select SND_SST_PLATFORM + help + This adds support for ASoC machine driver for Intel(R) MID Medfield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". + +config SND_SST_PLATFORM + tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile new file mode 100644 index 0000000..6398833 --- /dev/null +++ b/sound/soc/intel/Makefile @@ -0,0 +1,5 @@ +snd-soc-sst-platform-objs := sst_platform.o +snd-soc-mfld-machine-objs := mfld_machine.o + +obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c new file mode 100644 index 0000000..d3d4c32 --- /dev/null +++ b/sound/soc/intel/mfld_machine.c @@ -0,0 +1,427 @@ +/* + * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/sn95031.h" + +#define MID_MONO 1 +#define MID_STEREO 2 +#define MID_MAX_CAP 5 +#define MFLD_JACK_INSERT 0x04 + +enum soc_mic_bias_zones { + MFLD_MV_START = 0, + /* mic bias volutage range for Headphones*/ + MFLD_MV_HP = 400, + /* mic bias volutage range for American Headset*/ + MFLD_MV_AM_HS = 650, + /* mic bias volutage range for Headset*/ + MFLD_MV_HS = 2000, + MFLD_MV_UNDEFINED, +}; + +static unsigned int hs_switch; +static unsigned int lo_dac; + +struct mfld_mc_private { + void __iomem *int_base; + u8 interrupt_status; +}; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + +/* sound card controls */ +static const char *headset_switch_text[] = {"Earpiece", "Headset"}; + +static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; + +static const struct soc_enum headset_enum = + SOC_ENUM_SINGLE_EXT(2, headset_switch_text); + +static const struct soc_enum lo_enum = + SOC_ENUM_SINGLE_EXT(4, lo_text); + +static int headset_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hs_switch; + return 0; +} + +static int headset_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == hs_switch) + return 0; + + if (ucontrol->value.integer.value[0]) { + pr_debug("hs_set HS path\n"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + pr_debug("hs_set EP path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } + snd_soc_dapm_sync(&codec->dapm); + hs_switch = ucontrol->value.integer.value[0]; + + return 0; +} + +static void lo_enable_out_pins(struct snd_soc_codec *codec) +{ + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); + if (hs_switch) { + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } +} + +static int lo_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lo_dac; + return 0; +} + +static int lo_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == lo_dac) + return 0; + + /* we dont want to work with last state of lineout so just enable all + * pins and then disable pins not required + */ + lo_enable_out_pins(codec); + switch (ucontrol->value.integer.value[0]) { + case 0: + pr_debug("set vibra path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); + break; + + case 1: + pr_debug("set hs path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); + break; + + case 2: + pr_debug("set spkr path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); + break; + + case 3: + pr_debug("set null path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); + break; + } + snd_soc_dapm_sync(&codec->dapm); + lo_dac = ucontrol->value.integer.value[0]; + return 0; +} + +static const struct snd_kcontrol_new mfld_snd_controls[] = { + SOC_ENUM_EXT("Playback Switch", headset_enum, + headset_get_switch, headset_set_switch), + SOC_ENUM_EXT("Lineout Mux", lo_enum, + lo_get_switch, lo_set_switch), +}; + +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + +static void mfld_jack_check(unsigned int intr_status) +{ + struct mfld_jack_data jack_data; + + jack_data.mfld_jack = &mfld_jack; + jack_data.intr_id = intr_status; + + sn95031_jack_detection(&jack_data); + /* TODO: add american headset detection post gpiolib support */ +} + +static int mfld_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret_val; + + /* Add jack sense widgets */ + snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); + + /* Set up the map */ + snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + + ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, + ARRAY_SIZE(mfld_snd_controls)); + if (ret_val) { + pr_err("soc_add_controls failed %d", ret_val); + return ret_val; + } + /* default is earpiece pin, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "Headphones"); + /* default is lineout NC, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); + lo_dac = 3; + hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + + /* Headset and button jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } + + /* we want to check if anything is inserted at boot, + * so send a fake event to codec and it will read adc + * to find if anything is there or not */ + mfld_jack_check(MFLD_JACK_INSERT); + return ret_val; +} + +static struct snd_soc_dai_link mfld_msic_dailink[] = { + { + .name = "Medfield Headset", + .stream_name = "Headset", + .cpu_dai_name = "Headset-cpu-dai", + .codec_dai_name = "SN95031 Headset", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = mfld_init, + }, + { + .name = "Medfield Speaker", + .stream_name = "Speaker", + .cpu_dai_name = "Speaker-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Vibra", + .stream_name = "Vibra1", + .cpu_dai_name = "Vibra1-cpu-dai", + .codec_dai_name = "SN95031 Vibra1", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Haptics", + .stream_name = "Vibra2", + .cpu_dai_name = "Vibra2-cpu-dai", + .codec_dai_name = "SN95031 Vibra2", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Compress", + .stream_name = "Speaker", + .cpu_dai_name = "Compress-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mfld = { + .name = "medfield_audio", + .owner = THIS_MODULE, + .dai_link = mfld_msic_dailink, + .num_links = ARRAY_SIZE(mfld_msic_dailink), +}; + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + memcpy_fromio(&mc_private->interrupt_status, + ((void *)(mc_private->int_base)), + sizeof(u8)); + return IRQ_WAKE_THREAD; +} + +static irqreturn_t snd_mfld_jack_detection(int irq, void *data) +{ + struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; + + if (mfld_jack.codec == NULL) + return IRQ_HANDLED; + mfld_jack_check(mc_drv_ctx->interrupt_status); + + return IRQ_HANDLED; +} + +static int snd_mfld_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0, irq; + struct mfld_mc_private *mc_drv_ctx; + struct resource *irq_mem; + + pr_debug("snd_mfld_mc_probe called\n"); + + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); + if (!mc_drv_ctx) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + + irq_mem = platform_get_resource_byname( + pdev, IORESOURCE_MEM, "IRQ_BASE"); + if (!irq_mem) { + pr_err("no mem resource given\n"); + return -ENODEV; + } + mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, + resource_size(irq_mem)); + if (!mc_drv_ctx->int_base) { + pr_err("Mapping of cache failed\n"); + return -ENOMEM; + } + /* register for interrupt */ + ret_val = devm_request_threaded_irq(&pdev->dev, irq, + snd_mfld_jack_intr_handler, + snd_mfld_jack_detection, + IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + return ret_val; + } + /* register the soc card */ + snd_soc_card_mfld.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); + if (ret_val) { + pr_debug("snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, mc_drv_ctx); + pr_debug("successfully exited probe\n"); + return 0; +} + +static struct platform_driver snd_mfld_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "msic_audio", + }, + .probe = snd_mfld_mc_probe, +}; + +module_platform_driver(snd_mfld_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:msic-audio"); diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst_dsp.h new file mode 100644 index 0000000..0fce1de --- /dev/null +++ b/sound/soc/intel/sst_dsp.h @@ -0,0 +1,134 @@ +#ifndef __SST_DSP_H__ +#define __SST_DSP_H__ +/* + * sst_dsp.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-12 Intel Corporation + * Authors: Vinod Koul + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +enum sst_codec_types { + /* AUDIO/MUSIC CODEC Type Definitions */ + SST_CODEC_TYPE_UNKNOWN = 0, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, +}; + +enum stream_type { + SST_STREAM_TYPE_NONE = 0, + SST_STREAM_TYPE_MUSIC = 1, +}; + +struct snd_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u8 use_offload_path; + u8 reserved2; + u16 reserved3; + u8 channel_map[8]; +} __packed; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 crc_check; /* crc_check - disable (0) or enable (1) */ + u8 reserved1; /* unused*/ + u16 reserved2; /* Unused */ +} __packed; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo*/ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ + u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + u16 reser2; + u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ + u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ + u8 reser1; + u16 reser3; +} __packed; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 brate; /* Use the hard coded value. */ + u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + u32 channel_mask; /* Channel Mask */ + u16 format_tag; /* Format Tag */ + u16 block_align; /* packet size */ + u16 wma_encode_opt;/* Encoder option */ + u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ + u8 reserved; /* reserved */ +} __packed; + +/* Codec params struture */ +union snd_sst_codec_params { + struct snd_pcm_params pcm_params; + struct snd_mp3_params mp3_params; + struct snd_aac_params aac_params; + struct snd_wma_params wma_params; +} __packed; + +/* Address and size info of a frame buffer */ +struct sst_address_info { + u32 addr; /* Address at IA */ + u32 size; /* Size of the buffer */ +}; + +struct snd_sst_alloc_params_ext { + struct sst_address_info ring_buf_info[8]; + u8 sg_count; + u8 reserved; + u16 reserved2; + u32 frag_size; /*Number of samples after which period elapsed + message is sent valid only if path = 0*/ +} __packed; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; +} __packed; + +struct snd_sst_params { + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct snd_sst_stream_params sparams; + struct snd_sst_alloc_params_ext aparams; +}; + +#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c new file mode 100644 index 0000000..b6b5eb6 --- /dev/null +++ b/sound/soc/intel/sst_platform.c @@ -0,0 +1,735 @@ +/* + * sst_platform.c - Intel MID Platform driver + * + * Copyright (C) 2010-2013 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sst_platform.h" + +static struct sst_device *sst; +static DEFINE_MUTEX(sst_lock); + +int sst_register_dsp(struct sst_device *dev) +{ + if (WARN_ON(!dev)) + return -EINVAL; + if (!try_module_get(dev->dev->driver->owner)) + return -ENODEV; + mutex_lock(&sst_lock); + if (sst) { + pr_err("we already have a device %s\n", sst->name); + module_put(dev->dev->driver->owner); + mutex_unlock(&sst_lock); + return -EEXIST; + } + pr_debug("registering device %s\n", dev->name); + sst = dev; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_register_dsp); + +int sst_unregister_dsp(struct sst_device *dev) +{ + if (WARN_ON(!dev)) + return -EINVAL; + if (dev != sst) + return -EINVAL; + + mutex_lock(&sst_lock); + + if (!sst) { + mutex_unlock(&sst_lock); + return -EIO; + } + + module_put(sst->dev->driver->owner); + pr_debug("unreg %s\n", sst->name); + sst = NULL; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_unregister_dsp); + +static struct snd_pcm_hardware sst_platform_pcm_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | + SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = SST_MIN_RATE, + .rate_max = SST_MAX_RATE, + .channels_min = SST_MIN_CHANNEL, + .channels_max = SST_MAX_CHANNEL, + .buffer_bytes_max = SST_MAX_BUFFER, + .period_bytes_min = SST_MIN_PERIOD_BYTES, + .period_bytes_max = SST_MAX_PERIOD_BYTES, + .periods_min = SST_MIN_PERIODS, + .periods_max = SST_MAX_PERIODS, + .fifo_size = SST_FIFO_SIZE, +}; + +/* MFLD - MSIC */ +static struct snd_soc_dai_driver sst_platform_dai[] = { +{ + .name = "Headset-cpu-dai", + .id = 0, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Speaker-cpu-dai", + .id = 1, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra1-cpu-dai", + .id = 2, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_MONO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra2-cpu-dai", + .id = 3, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Compress-cpu-dai", + .compress_dai = 1, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +}; + +static const struct snd_soc_component_driver sst_component = { + .name = "sst", +}; + +/* helper functions */ +static inline void sst_set_stream_status(struct sst_runtime_stream *stream, + int state) +{ + unsigned long flags; + spin_lock_irqsave(&stream->status_lock, flags); + stream->stream_status = state; + spin_unlock_irqrestore(&stream->status_lock, flags); +} + +static inline int sst_get_stream_status(struct sst_runtime_stream *stream) +{ + int state; + unsigned long flags; + + spin_lock_irqsave(&stream->status_lock, flags); + state = stream->stream_status; + spin_unlock_irqrestore(&stream->status_lock, flags); + return state; +} + +static void sst_fill_pcm_params(struct snd_pcm_substream *substream, + struct sst_pcm_params *param) +{ + + param->codec = SST_CODEC_TYPE_PCM; + param->num_chan = (u8) substream->runtime->channels; + param->pcm_wd_sz = substream->runtime->sample_bits; + param->reserved = 0; + param->sfreq = substream->runtime->rate; + param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); + param->period_count = substream->runtime->period_size; + param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); +} + +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + struct sst_pcm_params param = {0}; + struct sst_stream_params str_params = {0}; + int ret_val; + + /* set codec params and inform SST driver the same */ + sst_fill_pcm_params(substream, ¶m); + substream->runtime->dma_area = substream->dma_buffer.area; + str_params.sparams = param; + str_params.codec = param.codec; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.device_type = substream->pcm->device + 1; + pr_debug("Playbck stream,Device %d\n", + substream->pcm->device); + } else { + str_params.ops = STREAM_OPS_CAPTURE; + str_params.device_type = SND_SST_DEVICE_CAPTURE; + pr_debug("Capture stream,Device %d\n", + substream->pcm->device); + } + ret_val = stream->ops->open(&str_params); + pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_info.str_id = ret_val; + pr_debug("str id : %d\n", stream->stream_info.str_id); + return ret_val; +} + +static void sst_period_elapsed(void *mad_substream) +{ + struct snd_pcm_substream *substream = mad_substream; + struct sst_runtime_stream *stream; + int status; + + if (!substream || !substream->runtime) + return; + stream = substream->runtime->private_data; + if (!stream) + return; + status = sst_get_stream_status(stream); + if (status != SST_PLATFORM_RUNNING) + return; + snd_pcm_period_elapsed(substream); +} + +static int sst_platform_init_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + int ret_val; + + pr_debug("setting buffer ptr param\n"); + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.period_elapsed = sst_period_elapsed; + stream->stream_info.mad_substream = substream; + stream->stream_info.buffer_ptr = 0; + stream->stream_info.sfreq = substream->runtime->rate; + ret_val = stream->ops->device_control( + SST_SND_STREAM_INIT, &stream->stream_info); + if (ret_val) + pr_err("control_set ret error %d\n", ret_val); + return ret_val; + +} +/* end -- helper functions */ + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sst_runtime_stream *stream; + int ret_val; + + pr_debug("sst_platform_open called\n"); + + snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + ret_val = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret_val < 0) + return ret_val; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + mutex_lock(&sst_lock); + if (!sst) { + pr_err("no device available to run\n"); + mutex_unlock(&sst_lock); + kfree(stream); + return -ENODEV; + } + if (!try_module_get(sst->dev->driver->owner)) { + mutex_unlock(&sst_lock); + kfree(stream); + return -ENODEV; + } + stream->ops = sst->ops; + mutex_unlock(&sst_lock); + + stream->stream_info.str_id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ + runtime->private_data = stream; + + return 0; +} + +static int sst_platform_close(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_close called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (str_id) + ret_val = stream->ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + return ret_val; +} + +static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_pcm_prepare called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (stream->stream_info.str_id) { + ret_val = stream->ops->device_control( + SST_SND_DROP, &str_id); + return ret_val; + } + + ret_val = sst_platform_alloc_stream(substream); + if (ret_val < 0) + return ret_val; + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_info.str_id); + + ret_val = sst_platform_init_stream(substream); + if (ret_val) + return ret_val; + substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; + return ret_val; +} + +static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, str_id; + struct sst_runtime_stream *stream; + int str_cmd, status; + + pr_debug("sst_platform_pcm_trigger called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("sst: Trigger Start\n"); + str_cmd = SST_SND_START; + status = SST_PLATFORM_RUNNING; + stream->stream_info.mad_substream = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("sst: in stop\n"); + str_cmd = SST_SND_DROP; + status = SST_PLATFORM_DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pr_debug("sst: in pause\n"); + str_cmd = SST_SND_PAUSE; + status = SST_PLATFORM_PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("sst: in pause release\n"); + str_cmd = SST_SND_RESUME; + status = SST_PLATFORM_RUNNING; + break; + default: + return -EINVAL; + } + ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) + sst_set_stream_status(stream, status); + + return ret_val; +} + + +static snd_pcm_uframes_t sst_platform_pcm_pointer + (struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val, status; + struct pcm_stream_info *str_info; + + stream = substream->runtime->private_data; + status = sst_get_stream_status(stream); + if (status == SST_PLATFORM_INIT) + return 0; + str_info = &stream->stream_info; + ret_val = stream->ops->device_control( + SST_SND_BUFFER_POINTER, str_info); + if (ret_val) { + pr_err("sst: error code = %d\n", ret_val); + return ret_val; + } + return stream->stream_info.buffer_ptr; +} + +static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + + return 0; +} + +static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_pcm_ops sst_platform_ops = { + .open = sst_platform_open, + .close = sst_platform_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = sst_platform_pcm_prepare, + .trigger = sst_platform_pcm_trigger, + .pointer = sst_platform_pcm_pointer, + .hw_params = sst_platform_pcm_hw_params, + .hw_free = sst_platform_pcm_hw_free, +}; + +static void sst_pcm_free(struct snd_pcm *pcm) +{ + pr_debug("sst_pcm_free called\n"); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + int retval = 0; + + pr_debug("sst_pcm_new called\n"); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + SST_MIN_BUFFER, SST_MAX_BUFFER); + if (retval) { + pr_err("dma buffer allocationf fail\n"); + return retval; + } + } + return retval; +} + +/* compress stream operations */ +static void sst_compr_fragment_elapsed(void *arg) +{ + struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; + + pr_debug("fragment elapsed by driver\n"); + if (cstream) + snd_compr_fragment_elapsed(cstream); +} + +static int sst_platform_compr_open(struct snd_compr_stream *cstream) +{ + + int ret_val = 0; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sst_runtime_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + if (!sst || !try_module_get(sst->dev->driver->owner)) { + pr_err("no device available to run\n"); + ret_val = -ENODEV; + goto out_ops; + } + stream->compr_ops = sst->compr_ops; + + stream->id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + runtime->private_data = stream; + return 0; +out_ops: + kfree(stream); + return ret_val; +} + +static int sst_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + stream = cstream->runtime->private_data; + /*need to check*/ + str_id = stream->id; + if (str_id) + ret_val = stream->compr_ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + pr_debug("%s: %d\n", __func__, ret_val); + return 0; +} + +static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct sst_runtime_stream *stream; + int retval; + struct snd_sst_params str_params; + struct sst_compress_cb cb; + + stream = cstream->runtime->private_data; + /* construct fw structure for this*/ + memset(&str_params, 0, sizeof(str_params)); + + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.stream_type = SST_STREAM_TYPE_MUSIC; + str_params.device_type = SND_SST_DEVICE_COMPRESS; + + switch (params->codec.id) { + case SND_AUDIOCODEC_MP3: { + str_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; + break; + } + + case SND_AUDIOCODEC_AAC: { + str_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.aac_params.pcm_wd_sz = 16; + if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_ADTS; + else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_RAW; + else { + pr_err("Undefined format%d\n", params->codec.format); + return -EINVAL; + } + str_params.sparams.uc.aac_params.externalsr = + params->codec.sample_rate; + break; + } + + default: + pr_err("codec not supported, id =%d\n", params->codec.id); + return -EINVAL; + } + + str_params.aparams.ring_buf_info[0].addr = + virt_to_phys(cstream->runtime->buffer); + str_params.aparams.ring_buf_info[0].size = + cstream->runtime->buffer_size; + str_params.aparams.sg_count = 1; + str_params.aparams.frag_size = cstream->runtime->fragment_size; + + cb.param = cstream; + cb.compr_cb = sst_compr_fragment_elapsed; + + retval = stream->compr_ops->open(&str_params, &cb); + if (retval < 0) { + pr_err("stream allocation failed %d\n", retval); + return retval; + } + + stream->id = retval; + return 0; +} + +static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->control(cmd, stream->id); +} + +static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->tstamp(stream->id, tstamp); + tstamp->byte_offset = tstamp->copied_total % + (u32)cstream->runtime->buffer_size; + pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); + return 0; +} + +static int sst_platform_compr_ack(struct snd_compr_stream *cstream, + size_t bytes) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->bytes_written += bytes; + + return 0; +} + +static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_caps(caps); +} + +static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_codec_caps(codec); +} + +static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->set_metadata(stream->id, metadata); +} + +static struct snd_compr_ops sst_platform_compr_ops = { + + .open = sst_platform_compr_open, + .free = sst_platform_compr_free, + .set_params = sst_platform_compr_set_params, + .set_metadata = sst_platform_compr_set_metadata, + .trigger = sst_platform_compr_trigger, + .pointer = sst_platform_compr_pointer, + .ack = sst_platform_compr_ack, + .get_caps = sst_platform_compr_get_caps, + .get_codec_caps = sst_platform_compr_get_codec_caps, +}; + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .ops = &sst_platform_ops, + .compr_ops = &sst_platform_compr_ops, + .pcm_new = sst_pcm_new, + .pcm_free = sst_pcm_free, +}; + +static int sst_platform_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("sst_platform_probe called\n"); + sst = NULL; + ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); + if (ret) { + pr_err("registering soc platform failed\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &sst_component, + sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); + if (ret) { + pr_err("registering cpu dais failed\n"); + snd_soc_unregister_platform(&pdev->dev); + } + return ret; +} + +static int sst_platform_remove(struct platform_device *pdev) +{ + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + pr_debug("sst_platform_remove success\n"); + return 0; +} + +static struct platform_driver sst_platform_driver = { + .driver = { + .name = "sst-platform", + .owner = THIS_MODULE, + }, + .probe = sst_platform_probe, + .remove = sst_platform_remove, +}; + +module_platform_driver(sst_platform_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sst-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h new file mode 100644 index 0000000..cacc906 --- /dev/null +++ b/sound/soc/intel/sst_platform.h @@ -0,0 +1,157 @@ +/* + * sst_platform.h - Intel MID Platform driver header file + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ + +#ifndef __SST_PLATFORMDRV_H__ +#define __SST_PLATFORMDRV_H__ + +#include "sst_dsp.h" + +#define SST_MONO 1 +#define SST_STEREO 2 +#define SST_MAX_CAP 5 + +#define SST_MIN_RATE 8000 +#define SST_MAX_RATE 48000 +#define SST_MIN_CHANNEL 1 +#define SST_MAX_CHANNEL 5 +#define SST_MAX_BUFFER (800*1024) +#define SST_MIN_BUFFER (800*1024) +#define SST_MIN_PERIOD_BYTES 32 +#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER +#define SST_MIN_PERIODS 2 +#define SST_MAX_PERIODS (1024*2) +#define SST_FIFO_SIZE 0 + +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long long buffer_ptr; + int sfreq; +}; + +enum sst_drv_status { + SST_PLATFORM_INIT = 1, + SST_PLATFORM_STARTED, + SST_PLATFORM_RUNNING, + SST_PLATFORM_PAUSED, + SST_PLATFORM_DROPPED, +}; + +enum sst_controls { + SST_SND_ALLOC = 0x00, + SST_SND_PAUSE = 0x01, + SST_SND_RESUME = 0x02, + SST_SND_DROP = 0x03, + SST_SND_FREE = 0x04, + SST_SND_BUFFER_POINTER = 0x05, + SST_SND_STREAM_INIT = 0x06, + SST_SND_START = 0x07, + SST_MAX_CONTROLS = 0x07, +}; + +enum sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, + STREAM_OPS_CAPTURE, +}; + +enum sst_audio_device_type { + SND_SST_DEVICE_HEADSET = 1, + SND_SST_DEVICE_IHF, + SND_SST_DEVICE_VIBRA, + SND_SST_DEVICE_HAPTIC, + SND_SST_DEVICE_CAPTURE, + SND_SST_DEVICE_COMPRESS, +}; + +/* PCM Parameters */ +struct sst_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u32 ring_buffer_size; + u32 period_count; /* period elapsed in samples*/ + u32 ring_buffer_addr; +}; + +struct sst_stream_params { + u32 result; + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct sst_pcm_params sparams; +}; + +struct sst_compress_cb { + void *param; + void (*compr_cb)(void *param); +}; + +struct compress_sst_ops { + const char *name; + int (*open) (struct snd_sst_params *str_params, + struct sst_compress_cb *cb); + int (*control) (unsigned int cmd, unsigned int str_id); + int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); + int (*ack) (unsigned int str_id, unsigned long bytes); + int (*close) (unsigned int str_id); + int (*get_caps) (struct snd_compr_caps *caps); + int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + int (*set_metadata) (unsigned int str_id, + struct snd_compr_metadata *mdata); + +}; + +struct sst_ops { + int (*open) (struct sst_stream_params *str_param); + int (*device_control) (int cmd, void *arg); + int (*close) (unsigned int str_id); +}; + +struct sst_runtime_stream { + int stream_status; + unsigned int id; + size_t bytes_written; + struct pcm_stream_info stream_info; + struct sst_ops *ops; + struct compress_sst_ops *compr_ops; + spinlock_t status_lock; +}; + +struct sst_device { + char *name; + struct device *dev; + struct sst_ops *ops; + struct compress_sst_ops *compr_ops; +}; + +int sst_register_dsp(struct sst_device *sst); +int sst_unregister_dsp(struct sst_device *sst); +#endif diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig deleted file mode 100644 index 61c10bf..0000000 --- a/sound/soc/mid-x86/Kconfig +++ /dev/null @@ -1,13 +0,0 @@ -config SND_MFLD_MACHINE - tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC - select SND_SOC_SN95031 - select SND_SST_PLATFORM - help - This adds support for ASoC machine driver for Intel(R) MID Medfield platform - used as alsa device in audio substem in Intel(R) MID devices - Say Y if you have such a device - If unsure select "N". - -config SND_SST_PLATFORM - tristate diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/mid-x86/Makefile deleted file mode 100644 index 6398833..0000000 --- a/sound/soc/mid-x86/Makefile +++ /dev/null @@ -1,5 +0,0 @@ -snd-soc-sst-platform-objs := sst_platform.o -snd-soc-mfld-machine-objs := mfld_machine.o - -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o -obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c deleted file mode 100644 index d3d4c32..0000000 --- a/sound/soc/mid-x86/mfld_machine.c +++ /dev/null @@ -1,427 +0,0 @@ -/* - * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/sn95031.h" - -#define MID_MONO 1 -#define MID_STEREO 2 -#define MID_MAX_CAP 5 -#define MFLD_JACK_INSERT 0x04 - -enum soc_mic_bias_zones { - MFLD_MV_START = 0, - /* mic bias volutage range for Headphones*/ - MFLD_MV_HP = 400, - /* mic bias volutage range for American Headset*/ - MFLD_MV_AM_HS = 650, - /* mic bias volutage range for Headset*/ - MFLD_MV_HS = 2000, - MFLD_MV_UNDEFINED, -}; - -static unsigned int hs_switch; -static unsigned int lo_dac; - -struct mfld_mc_private { - void __iomem *int_base; - u8 interrupt_status; -}; - -struct snd_soc_jack mfld_jack; - -/*Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin mfld_jack_pins[] = { - { - .pin = "Headphones", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "AMIC1", - .mask = SND_JACK_MICROPHONE, - }, -}; - -/* jack detection voltage zones */ -static struct snd_soc_jack_zone mfld_zones[] = { - {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, - {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, -}; - -/* sound card controls */ -static const char *headset_switch_text[] = {"Earpiece", "Headset"}; - -static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; - -static const struct soc_enum headset_enum = - SOC_ENUM_SINGLE_EXT(2, headset_switch_text); - -static const struct soc_enum lo_enum = - SOC_ENUM_SINGLE_EXT(4, lo_text); - -static int headset_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = hs_switch; - return 0; -} - -static int headset_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - if (ucontrol->value.integer.value[0] == hs_switch) - return 0; - - if (ucontrol->value.integer.value[0]) { - pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - } else { - pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); - } - snd_soc_dapm_sync(&codec->dapm); - hs_switch = ucontrol->value.integer.value[0]; - - return 0; -} - -static void lo_enable_out_pins(struct snd_soc_codec *codec) -{ - snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); - snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); - snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); - snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); - if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - } else { - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); - } -} - -static int lo_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lo_dac; - return 0; -} - -static int lo_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - if (ucontrol->value.integer.value[0] == lo_dac) - return 0; - - /* we dont want to work with last state of lineout so just enable all - * pins and then disable pins not required - */ - lo_enable_out_pins(codec); - switch (ucontrol->value.integer.value[0]) { - case 0: - pr_debug("set vibra path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); - snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); - break; - - case 1: - pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); - break; - - case 2: - pr_debug("set spkr path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); - break; - - case 3: - pr_debug("set null path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); - break; - } - snd_soc_dapm_sync(&codec->dapm); - lo_dac = ucontrol->value.integer.value[0]; - return 0; -} - -static const struct snd_kcontrol_new mfld_snd_controls[] = { - SOC_ENUM_EXT("Playback Switch", headset_enum, - headset_get_switch, headset_set_switch), - SOC_ENUM_EXT("Lineout Mux", lo_enum, - lo_get_switch, lo_set_switch), -}; - -static const struct snd_soc_dapm_widget mfld_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route mfld_map[] = { - {"Headphones", NULL, "HPOUTR"}, - {"Headphones", NULL, "HPOUTL"}, - {"Mic", NULL, "AMIC1"}, -}; - -static void mfld_jack_check(unsigned int intr_status) -{ - struct mfld_jack_data jack_data; - - jack_data.mfld_jack = &mfld_jack; - jack_data.intr_id = intr_status; - - sn95031_jack_detection(&jack_data); - /* TODO: add american headset detection post gpiolib support */ -} - -static int mfld_init(struct snd_soc_pcm_runtime *runtime) -{ - struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret_val; - - /* Add jack sense widgets */ - snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); - - /* Set up the map */ - snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); - snd_soc_dapm_enable_pin(dapm, "Mic"); - - ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, - ARRAY_SIZE(mfld_snd_controls)); - if (ret_val) { - pr_err("soc_add_controls failed %d", ret_val); - return ret_val; - } - /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "Headphones"); - /* default is lineout NC, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); - lo_dac = 3; - hs_switch = 0; - /* we dont use linein in this so set to NC */ - snd_soc_dapm_disable_pin(dapm, "LINEINL"); - snd_soc_dapm_disable_pin(dapm, "LINEINR"); - - /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); - if (ret_val) { - pr_err("jack creation failed\n"); - return ret_val; - } - - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } - ret_val = snd_soc_jack_add_zones(&mfld_jack, - ARRAY_SIZE(mfld_zones), mfld_zones); - if (ret_val) { - pr_err("adding jack zones failed\n"); - return ret_val; - } - - /* we want to check if anything is inserted at boot, - * so send a fake event to codec and it will read adc - * to find if anything is there or not */ - mfld_jack_check(MFLD_JACK_INSERT); - return ret_val; -} - -static struct snd_soc_dai_link mfld_msic_dailink[] = { - { - .name = "Medfield Headset", - .stream_name = "Headset", - .cpu_dai_name = "Headset-cpu-dai", - .codec_dai_name = "SN95031 Headset", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = mfld_init, - }, - { - .name = "Medfield Speaker", - .stream_name = "Speaker", - .cpu_dai_name = "Speaker-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Vibra", - .stream_name = "Vibra1", - .cpu_dai_name = "Vibra1-cpu-dai", - .codec_dai_name = "SN95031 Vibra1", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Haptics", - .stream_name = "Vibra2", - .cpu_dai_name = "Vibra2-cpu-dai", - .codec_dai_name = "SN95031 Vibra2", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Compress", - .stream_name = "Speaker", - .cpu_dai_name = "Compress-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, -}; - -/* SoC card */ -static struct snd_soc_card snd_soc_card_mfld = { - .name = "medfield_audio", - .owner = THIS_MODULE, - .dai_link = mfld_msic_dailink, - .num_links = ARRAY_SIZE(mfld_msic_dailink), -}; - -static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) -{ - struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; - - memcpy_fromio(&mc_private->interrupt_status, - ((void *)(mc_private->int_base)), - sizeof(u8)); - return IRQ_WAKE_THREAD; -} - -static irqreturn_t snd_mfld_jack_detection(int irq, void *data) -{ - struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; - mfld_jack_check(mc_drv_ctx->interrupt_status); - - return IRQ_HANDLED; -} - -static int snd_mfld_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0, irq; - struct mfld_mc_private *mc_drv_ctx; - struct resource *irq_mem; - - pr_debug("snd_mfld_mc_probe called\n"); - - /* retrive the irq number */ - irq = platform_get_irq(pdev, 0); - - /* audio interrupt base of SRAM location where - * interrupts are stored by System FW */ - mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); - if (!mc_drv_ctx) { - pr_err("allocation failed\n"); - return -ENOMEM; - } - - irq_mem = platform_get_resource_byname( - pdev, IORESOURCE_MEM, "IRQ_BASE"); - if (!irq_mem) { - pr_err("no mem resource given\n"); - return -ENODEV; - } - mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, - resource_size(irq_mem)); - if (!mc_drv_ctx->int_base) { - pr_err("Mapping of cache failed\n"); - return -ENOMEM; - } - /* register for interrupt */ - ret_val = devm_request_threaded_irq(&pdev->dev, irq, - snd_mfld_jack_intr_handler, - snd_mfld_jack_detection, - IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); - if (ret_val) { - pr_err("cannot register IRQ\n"); - return ret_val; - } - /* register the soc card */ - snd_soc_card_mfld.dev = &pdev->dev; - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); - if (ret_val) { - pr_debug("snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, mc_drv_ctx); - pr_debug("successfully exited probe\n"); - return 0; -} - -static struct platform_driver snd_mfld_mc_driver = { - .driver = { - .owner = THIS_MODULE, - .name = "msic_audio", - }, - .probe = snd_mfld_mc_probe, -}; - -module_platform_driver(snd_mfld_mc_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); -MODULE_AUTHOR("Vinod Koul "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:msic-audio"); diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h deleted file mode 100644 index 0fce1de..0000000 --- a/sound/soc/mid-x86/sst_dsp.h +++ /dev/null @@ -1,134 +0,0 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ -/* - * sst_dsp.h - Intel SST Driver for audio engine - * - * Copyright (C) 2008-12 Intel Corporation - * Authors: Vinod Koul - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -enum sst_codec_types { - /* AUDIO/MUSIC CODEC Type Definitions */ - SST_CODEC_TYPE_UNKNOWN = 0, - SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ - SST_CODEC_TYPE_MP3, - SST_CODEC_TYPE_MP24, - SST_CODEC_TYPE_AAC, - SST_CODEC_TYPE_AACP, - SST_CODEC_TYPE_eAACP, -}; - -enum stream_type { - SST_STREAM_TYPE_NONE = 0, - SST_STREAM_TYPE_MUSIC = 1, -}; - -struct snd_pcm_params { - u16 codec; /* codec type */ - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u8 use_offload_path; - u8 reserved2; - u16 reserved3; - u8 channel_map[8]; -} __packed; - -/* MP3 Music Parameters Message */ -struct snd_mp3_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u8 crc_check; /* crc_check - disable (0) or enable (1) */ - u8 reserved1; /* unused*/ - u16 reserved2; /* Unused */ -} __packed; - -#define AAC_BIT_STREAM_ADTS 0 -#define AAC_BIT_STREAM_ADIF 1 -#define AAC_BIT_STREAM_RAW 2 - -/* AAC Music Parameters Message */ -struct snd_aac_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo*/ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ - u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ - u16 reser2; - u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ - u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ - u8 reser1; - u16 reser3; -} __packed; - -/* WMA Music Parameters Message */ -struct snd_wma_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 brate; /* Use the hard coded value. */ - u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ - u32 channel_mask; /* Channel Mask */ - u16 format_tag; /* Format Tag */ - u16 block_align; /* packet size */ - u16 wma_encode_opt;/* Encoder option */ - u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ - u8 reserved; /* reserved */ -} __packed; - -/* Codec params struture */ -union snd_sst_codec_params { - struct snd_pcm_params pcm_params; - struct snd_mp3_params mp3_params; - struct snd_aac_params aac_params; - struct snd_wma_params wma_params; -} __packed; - -/* Address and size info of a frame buffer */ -struct sst_address_info { - u32 addr; /* Address at IA */ - u32 size; /* Size of the buffer */ -}; - -struct snd_sst_alloc_params_ext { - struct sst_address_info ring_buf_info[8]; - u8 sg_count; - u8 reserved; - u16 reserved2; - u32 frag_size; /*Number of samples after which period elapsed - message is sent valid only if path = 0*/ -} __packed; - -struct snd_sst_stream_params { - union snd_sst_codec_params uc; -} __packed; - -struct snd_sst_params { - u32 stream_id; - u8 codec; - u8 ops; - u8 stream_type; - u8 device_type; - struct snd_sst_stream_params sparams; - struct snd_sst_alloc_params_ext aparams; -}; - -#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c deleted file mode 100644 index b6b5eb6..0000000 --- a/sound/soc/mid-x86/sst_platform.c +++ /dev/null @@ -1,735 +0,0 @@ -/* - * sst_platform.c - Intel MID Platform driver - * - * Copyright (C) 2010-2013 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include -#include -#include -#include -#include -#include -#include -#include -#include "sst_platform.h" - -static struct sst_device *sst; -static DEFINE_MUTEX(sst_lock); - -int sst_register_dsp(struct sst_device *dev) -{ - if (WARN_ON(!dev)) - return -EINVAL; - if (!try_module_get(dev->dev->driver->owner)) - return -ENODEV; - mutex_lock(&sst_lock); - if (sst) { - pr_err("we already have a device %s\n", sst->name); - module_put(dev->dev->driver->owner); - mutex_unlock(&sst_lock); - return -EEXIST; - } - pr_debug("registering device %s\n", dev->name); - sst = dev; - mutex_unlock(&sst_lock); - return 0; -} -EXPORT_SYMBOL_GPL(sst_register_dsp); - -int sst_unregister_dsp(struct sst_device *dev) -{ - if (WARN_ON(!dev)) - return -EINVAL; - if (dev != sst) - return -EINVAL; - - mutex_lock(&sst_lock); - - if (!sst) { - mutex_unlock(&sst_lock); - return -EIO; - } - - module_put(sst->dev->driver->owner); - pr_debug("unreg %s\n", sst->name); - sst = NULL; - mutex_unlock(&sst_lock); - return 0; -} -EXPORT_SYMBOL_GPL(sst_unregister_dsp); - -static struct snd_pcm_hardware sst_platform_pcm_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_DOUBLE | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP| - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | - SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), - .rates = (SNDRV_PCM_RATE_8000| - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000), - .rate_min = SST_MIN_RATE, - .rate_max = SST_MAX_RATE, - .channels_min = SST_MIN_CHANNEL, - .channels_max = SST_MAX_CHANNEL, - .buffer_bytes_max = SST_MAX_BUFFER, - .period_bytes_min = SST_MIN_PERIOD_BYTES, - .period_bytes_max = SST_MAX_PERIOD_BYTES, - .periods_min = SST_MIN_PERIODS, - .periods_max = SST_MAX_PERIODS, - .fifo_size = SST_FIFO_SIZE, -}; - -/* MFLD - MSIC */ -static struct snd_soc_dai_driver sst_platform_dai[] = { -{ - .name = "Headset-cpu-dai", - .id = 0, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 5, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Speaker-cpu-dai", - .id = 1, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Vibra1-cpu-dai", - .id = 2, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_MONO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Vibra2-cpu-dai", - .id = 3, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Compress-cpu-dai", - .compress_dai = 1, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -}; - -static const struct snd_soc_component_driver sst_component = { - .name = "sst", -}; - -/* helper functions */ -static inline void sst_set_stream_status(struct sst_runtime_stream *stream, - int state) -{ - unsigned long flags; - spin_lock_irqsave(&stream->status_lock, flags); - stream->stream_status = state; - spin_unlock_irqrestore(&stream->status_lock, flags); -} - -static inline int sst_get_stream_status(struct sst_runtime_stream *stream) -{ - int state; - unsigned long flags; - - spin_lock_irqsave(&stream->status_lock, flags); - state = stream->stream_status; - spin_unlock_irqrestore(&stream->status_lock, flags); - return state; -} - -static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) -{ - - param->codec = SST_CODEC_TYPE_PCM; - param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); -} - -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream = - substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; - - /* set codec params and inform SST driver the same */ - sst_fill_pcm_params(substream, ¶m); - substream->runtime->dma_area = substream->dma_buffer.area; - str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); - if (ret_val < 0) - return ret_val; - - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); - return ret_val; -} - -static void sst_period_elapsed(void *mad_substream) -{ - struct snd_pcm_substream *substream = mad_substream; - struct sst_runtime_stream *stream; - int status; - - if (!substream || !substream->runtime) - return; - stream = substream->runtime->private_data; - if (!stream) - return; - status = sst_get_stream_status(stream); - if (status != SST_PLATFORM_RUNNING) - return; - snd_pcm_period_elapsed(substream); -} - -static int sst_platform_init_stream(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream = - substream->runtime->private_data; - int ret_val; - - pr_debug("setting buffer ptr param\n"); - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; - stream->stream_info.buffer_ptr = 0; - stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); - if (ret_val) - pr_err("control_set ret error %d\n", ret_val); - return ret_val; - -} -/* end -- helper functions */ - -static int sst_platform_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; - - stream = kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) - return -ENOMEM; - spin_lock_init(&stream->status_lock); - - /* get the sst ops */ - mutex_lock(&sst_lock); - if (!sst) { - pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - stream->ops = sst->ops; - mutex_unlock(&sst_lock); - - stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; - /* allocate memory for SST API set */ - runtime->private_data = stream; - - return 0; -} - -static int sst_platform_close(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - pr_debug("sst_platform_close called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - if (str_id) - ret_val = stream->ops->close(str_id); - module_put(sst->dev->driver->owner); - kfree(stream); - return ret_val; -} - -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - pr_debug("sst_platform_pcm_prepare called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - if (stream->stream_info.str_id) { - ret_val = stream->ops->device_control( - SST_SND_DROP, &str_id); - return ret_val; - } - - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) - return ret_val; - snprintf(substream->pcm->id, sizeof(substream->pcm->id), - "%d", stream->stream_info.str_id); - - ret_val = sst_platform_init_stream(substream); - if (ret_val) - return ret_val; - substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; - return ret_val; -} - -static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - int ret_val = 0, str_id; - struct sst_runtime_stream *stream; - int str_cmd, status; - - pr_debug("sst_platform_pcm_trigger called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pr_debug("sst: Trigger Start\n"); - str_cmd = SST_SND_START; - status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; - status = SST_PLATFORM_DROPPED; - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; - status = SST_PLATFORM_PAUSED; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; - status = SST_PLATFORM_RUNNING; - break; - default: - return -EINVAL; - } - ret_val = stream->ops->device_control(str_cmd, &str_id); - if (!ret_val) - sst_set_stream_status(stream, status); - - return ret_val; -} - - -static snd_pcm_uframes_t sst_platform_pcm_pointer - (struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val, status; - struct pcm_stream_info *str_info; - - stream = substream->runtime->private_data; - status = sst_get_stream_status(stream); - if (status == SST_PLATFORM_INIT) - return 0; - str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); - if (ret_val) { - pr_err("sst: error code = %d\n", ret_val); - return ret_val; - } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static struct snd_pcm_ops sst_platform_ops = { - .open = sst_platform_open, - .close = sst_platform_close, - .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, - .trigger = sst_platform_pcm_trigger, - .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, -}; - -static void sst_pcm_free(struct snd_pcm *pcm) -{ - pr_debug("sst_pcm_free called\n"); - snd_pcm_lib_preallocate_free_for_all(pcm); -} - -static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_pcm *pcm = rtd->pcm; - int retval = 0; - - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - retval = snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - SST_MIN_BUFFER, SST_MAX_BUFFER); - if (retval) { - pr_err("dma buffer allocationf fail\n"); - return retval; - } - } - return retval; -} - -/* compress stream operations */ -static void sst_compr_fragment_elapsed(void *arg) -{ - struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; - - pr_debug("fragment elapsed by driver\n"); - if (cstream) - snd_compr_fragment_elapsed(cstream); -} - -static int sst_platform_compr_open(struct snd_compr_stream *cstream) -{ - - int ret_val = 0; - struct snd_compr_runtime *runtime = cstream->runtime; - struct sst_runtime_stream *stream; - - stream = kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) - return -ENOMEM; - - spin_lock_init(&stream->status_lock); - - /* get the sst ops */ - if (!sst || !try_module_get(sst->dev->driver->owner)) { - pr_err("no device available to run\n"); - ret_val = -ENODEV; - goto out_ops; - } - stream->compr_ops = sst->compr_ops; - - stream->id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - runtime->private_data = stream; - return 0; -out_ops: - kfree(stream); - return ret_val; -} - -static int sst_platform_compr_free(struct snd_compr_stream *cstream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - stream = cstream->runtime->private_data; - /*need to check*/ - str_id = stream->id; - if (str_id) - ret_val = stream->compr_ops->close(str_id); - module_put(sst->dev->driver->owner); - kfree(stream); - pr_debug("%s: %d\n", __func__, ret_val); - return 0; -} - -static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, - struct snd_compr_params *params) -{ - struct sst_runtime_stream *stream; - int retval; - struct snd_sst_params str_params; - struct sst_compress_cb cb; - - stream = cstream->runtime->private_data; - /* construct fw structure for this*/ - memset(&str_params, 0, sizeof(str_params)); - - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; - - switch (params->codec.id) { - case SND_AUDIOCODEC_MP3: { - str_params.codec = SST_CODEC_TYPE_MP3; - str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; - str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; - str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; - break; - } - - case SND_AUDIOCODEC_AAC: { - str_params.codec = SST_CODEC_TYPE_AAC; - str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; - str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; - str_params.sparams.uc.aac_params.pcm_wd_sz = 16; - if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) - str_params.sparams.uc.aac_params.bs_format = - AAC_BIT_STREAM_ADTS; - else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) - str_params.sparams.uc.aac_params.bs_format = - AAC_BIT_STREAM_RAW; - else { - pr_err("Undefined format%d\n", params->codec.format); - return -EINVAL; - } - str_params.sparams.uc.aac_params.externalsr = - params->codec.sample_rate; - break; - } - - default: - pr_err("codec not supported, id =%d\n", params->codec.id); - return -EINVAL; - } - - str_params.aparams.ring_buf_info[0].addr = - virt_to_phys(cstream->runtime->buffer); - str_params.aparams.ring_buf_info[0].size = - cstream->runtime->buffer_size; - str_params.aparams.sg_count = 1; - str_params.aparams.frag_size = cstream->runtime->fragment_size; - - cb.param = cstream; - cb.compr_cb = sst_compr_fragment_elapsed; - - retval = stream->compr_ops->open(&str_params, &cb); - if (retval < 0) { - pr_err("stream allocation failed %d\n", retval); - return retval; - } - - stream->id = retval; - return 0; -} - -static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->control(cmd, stream->id); -} - -static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, - struct snd_compr_tstamp *tstamp) -{ - struct sst_runtime_stream *stream; - - stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); - tstamp->byte_offset = tstamp->copied_total % - (u32)cstream->runtime->buffer_size; - pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); - return 0; -} - -static int sst_platform_compr_ack(struct snd_compr_stream *cstream, - size_t bytes) -{ - struct sst_runtime_stream *stream; - - stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); - stream->bytes_written += bytes; - - return 0; -} - -static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, - struct snd_compr_caps *caps) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->get_caps(caps); -} - -static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, - struct snd_compr_codec_caps *codec) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->get_codec_caps(codec); -} - -static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, - struct snd_compr_metadata *metadata) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->set_metadata(stream->id, metadata); -} - -static struct snd_compr_ops sst_platform_compr_ops = { - - .open = sst_platform_compr_open, - .free = sst_platform_compr_free, - .set_params = sst_platform_compr_set_params, - .set_metadata = sst_platform_compr_set_metadata, - .trigger = sst_platform_compr_trigger, - .pointer = sst_platform_compr_pointer, - .ack = sst_platform_compr_ack, - .get_caps = sst_platform_compr_get_caps, - .get_codec_caps = sst_platform_compr_get_codec_caps, -}; - -static struct snd_soc_platform_driver sst_soc_platform_drv = { - .ops = &sst_platform_ops, - .compr_ops = &sst_platform_compr_ops, - .pcm_new = sst_pcm_new, - .pcm_free = sst_pcm_free, -}; - -static int sst_platform_probe(struct platform_device *pdev) -{ - int ret; - - pr_debug("sst_platform_probe called\n"); - sst = NULL; - ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); - if (ret) { - pr_err("registering soc platform failed\n"); - return ret; - } - - ret = snd_soc_register_component(&pdev->dev, &sst_component, - sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); - if (ret) { - pr_err("registering cpu dais failed\n"); - snd_soc_unregister_platform(&pdev->dev); - } - return ret; -} - -static int sst_platform_remove(struct platform_device *pdev) -{ - - snd_soc_unregister_component(&pdev->dev); - snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove success\n"); - return 0; -} - -static struct platform_driver sst_platform_driver = { - .driver = { - .name = "sst-platform", - .owner = THIS_MODULE, - }, - .probe = sst_platform_probe, - .remove = sst_platform_remove, -}; - -module_platform_driver(sst_platform_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); -MODULE_AUTHOR("Vinod Koul "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h deleted file mode 100644 index cacc906..0000000 --- a/sound/soc/mid-x86/sst_platform.h +++ /dev/null @@ -1,157 +0,0 @@ -/* - * sst_platform.h - Intel MID Platform driver header file - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ - -#ifndef __SST_PLATFORMDRV_H__ -#define __SST_PLATFORMDRV_H__ - -#include "sst_dsp.h" - -#define SST_MONO 1 -#define SST_STEREO 2 -#define SST_MAX_CAP 5 - -#define SST_MIN_RATE 8000 -#define SST_MAX_RATE 48000 -#define SST_MIN_CHANNEL 1 -#define SST_MAX_CHANNEL 5 -#define SST_MAX_BUFFER (800*1024) -#define SST_MIN_BUFFER (800*1024) -#define SST_MIN_PERIOD_BYTES 32 -#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER -#define SST_MIN_PERIODS 2 -#define SST_MAX_PERIODS (1024*2) -#define SST_FIFO_SIZE 0 - -struct pcm_stream_info { - int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); - unsigned long long buffer_ptr; - int sfreq; -}; - -enum sst_drv_status { - SST_PLATFORM_INIT = 1, - SST_PLATFORM_STARTED, - SST_PLATFORM_RUNNING, - SST_PLATFORM_PAUSED, - SST_PLATFORM_DROPPED, -}; - -enum sst_controls { - SST_SND_ALLOC = 0x00, - SST_SND_PAUSE = 0x01, - SST_SND_RESUME = 0x02, - SST_SND_DROP = 0x03, - SST_SND_FREE = 0x04, - SST_SND_BUFFER_POINTER = 0x05, - SST_SND_STREAM_INIT = 0x06, - SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, -}; - -enum sst_stream_ops { - STREAM_OPS_PLAYBACK = 0, - STREAM_OPS_CAPTURE, -}; - -enum sst_audio_device_type { - SND_SST_DEVICE_HEADSET = 1, - SND_SST_DEVICE_IHF, - SND_SST_DEVICE_VIBRA, - SND_SST_DEVICE_HAPTIC, - SND_SST_DEVICE_CAPTURE, - SND_SST_DEVICE_COMPRESS, -}; - -/* PCM Parameters */ -struct sst_pcm_params { - u16 codec; /* codec type */ - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u32 ring_buffer_size; - u32 period_count; /* period elapsed in samples*/ - u32 ring_buffer_addr; -}; - -struct sst_stream_params { - u32 result; - u32 stream_id; - u8 codec; - u8 ops; - u8 stream_type; - u8 device_type; - struct sst_pcm_params sparams; -}; - -struct sst_compress_cb { - void *param; - void (*compr_cb)(void *param); -}; - -struct compress_sst_ops { - const char *name; - int (*open) (struct snd_sst_params *str_params, - struct sst_compress_cb *cb); - int (*control) (unsigned int cmd, unsigned int str_id); - int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); - int (*ack) (unsigned int str_id, unsigned long bytes); - int (*close) (unsigned int str_id); - int (*get_caps) (struct snd_compr_caps *caps); - int (*get_codec_caps) (struct snd_compr_codec_caps *codec); - int (*set_metadata) (unsigned int str_id, - struct snd_compr_metadata *mdata); - -}; - -struct sst_ops { - int (*open) (struct sst_stream_params *str_param); - int (*device_control) (int cmd, void *arg); - int (*close) (unsigned int str_id); -}; - -struct sst_runtime_stream { - int stream_status; - unsigned int id; - size_t bytes_written; - struct pcm_stream_info stream_info; - struct sst_ops *ops; - struct compress_sst_ops *compr_ops; - spinlock_t status_lock; -}; - -struct sst_device { - char *name; - struct device *dev; - struct sst_ops *ops; - struct compress_sst_ops *compr_ops; -}; - -int sst_register_dsp(struct sst_device *sst); -int sst_unregister_dsp(struct sst_device *sst); -#endif -- cgit v0.10.2 From 8e6ad35a31e7ebc59543df875fc970200df2cf68 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:51 -0200 Subject: ASoC: wm8510: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6ed5433..7df7d45 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -684,7 +684,7 @@ static struct spi_driver wm8510_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8510_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -735,7 +735,7 @@ static struct i2c_driver wm8510_i2c_driver = { static int __init wm8510_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8510_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8510 I2C driver: %d\n", @@ -755,7 +755,7 @@ module_init(wm8510_modinit); static void __exit wm8510_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8510_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 008ef947d0c5d14442256a37f6bf6b14015efe26 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:50 -0200 Subject: ASoC: wm8523: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 139bf9a..74d106d 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -452,7 +452,7 @@ static const struct regmap_config wm8523_regmap = { .volatile_reg = wm8523_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8523_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -555,7 +555,7 @@ static struct i2c_driver wm8523_i2c_driver = { static int __init wm8523_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8523_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8523 I2C driver: %d\n", @@ -568,7 +568,7 @@ module_init(wm8523_modinit); static void __exit wm8523_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8523_i2c_driver); #endif } -- cgit v0.10.2 From f58c4fc4a3bf9eb699a638634d20ef4d16069366 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:49 -0200 Subject: ASoC: wm8580: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 08a414b..318989a 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -941,7 +941,7 @@ static const struct regmap_config wm8580_regmap = { .volatile_reg = wm8580_volatile, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1003,7 +1003,7 @@ static int __init wm8580_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8580_i2c_driver); if (ret != 0) { pr_err("Failed to register WM8580 I2C driver: %d\n", ret); @@ -1016,7 +1016,7 @@ module_init(wm8580_modinit); static void __exit wm8580_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8580_i2c_driver); #endif } -- cgit v0.10.2 From 2309d6757900c4a6909fa673724976935b408a25 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:48 -0200 Subject: ASoC: wm8711: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 5b428b0..d99f948 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -469,7 +469,7 @@ static struct spi_driver wm8711_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8711_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -520,7 +520,7 @@ static struct i2c_driver wm8711_i2c_driver = { static int __init wm8711_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8711_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n", @@ -540,7 +540,7 @@ module_init(wm8711_modinit); static void __exit wm8711_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8711_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 5c1537163ce716e317776565b8210ea06fa2b5de Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:47 -0200 Subject: ASoC: wm8728: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index c6a292d..cd89033 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -320,7 +320,7 @@ static struct spi_driver wm8728_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8728_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -371,7 +371,7 @@ static struct i2c_driver wm8728_i2c_driver = { static int __init wm8728_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8728_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8728 I2C driver: %d\n", @@ -391,7 +391,7 @@ module_init(wm8728_modinit); static void __exit wm8728_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8728_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 6435e5be652633b87f73460f9c17be361404ee01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Nov 2013 17:19:55 +0000 Subject: ASoC: wm8940: Convert to table based control and DAPM init Signed-off-by: Mark Brown Acked-by: Charles Keepax Acked-by: Jonathan Cameron diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b1591c61..4858b5c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -264,7 +264,7 @@ static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AUX"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8940_dapm_routes[] = { /* Mono output mixer */ {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, @@ -296,21 +296,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ADC", NULL, "Boost Mixer"}, }; -static int wm8940_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, wm8940_dapm_widgets, - ARRAY_SIZE(wm8940_dapm_widgets)); - if (ret) - goto error_ret; - ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - -error_ret: - return ret; -} - #define wm8940_reset(c) snd_soc_write(c, WM8940_SOFTRESET, 0); static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, @@ -716,11 +701,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_codec_controls(codec, wm8940_snd_controls, - ARRAY_SIZE(wm8940_snd_controls)); - if (ret) - return ret; - ret = wm8940_add_widgets(codec); return ret; } @@ -736,6 +716,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .suspend = wm8940_suspend, .resume = wm8940_resume, .set_bias_level = wm8940_set_bias_level, + .controls = wm8940_snd_controls, + .num_controls = ARRAY_SIZE(wm8940_snd_controls), + .dapm_widgets = wm8940_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8940_dapm_widgets), + .dapm_routes = wm8940_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8940_dapm_routes), .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8940_reg_defaults, -- cgit v0.10.2 From fbbf7fea8e806ccc3ce0e059ad1d9671d57b4309 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Nov 2013 17:22:23 +0000 Subject: ASoC: wm8940: Convert to direct regmap API usage This helps move us towards being able to remove the ASoC level I/O code. Signed-off-by: Mark Brown Acked-by: Charles Keepax Acked-by: Jonathan Cameron diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 4858b5c..b404c26 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include #include #include @@ -41,78 +41,116 @@ struct wm8940_priv { unsigned int sysclk; - enum snd_soc_control_type control_type; + struct regmap *regmap; }; -static int wm8940_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8940_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8940_SOFTRESET: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8940_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8940_SOFTRESET: + case WM8940_POWER1: + case WM8940_POWER2: + case WM8940_POWER3: + case WM8940_IFACE: + case WM8940_COMPANDINGCTL: + case WM8940_CLOCK: + case WM8940_ADDCNTRL: + case WM8940_GPIO: + case WM8940_CTLINT: + case WM8940_DAC: + case WM8940_DACVOL: + case WM8940_ADC: + case WM8940_ADCVOL: + case WM8940_NOTCH1: + case WM8940_NOTCH2: + case WM8940_NOTCH3: + case WM8940_NOTCH4: + case WM8940_NOTCH5: + case WM8940_NOTCH6: + case WM8940_NOTCH7: + case WM8940_NOTCH8: + case WM8940_DACLIM1: + case WM8940_DACLIM2: + case WM8940_ALC1: + case WM8940_ALC2: + case WM8940_ALC3: + case WM8940_NOISEGATE: + case WM8940_PLLN: + case WM8940_PLLK1: + case WM8940_PLLK2: + case WM8940_PLLK3: + case WM8940_ALC4: + case WM8940_INPUTCTL: + case WM8940_PGAGAIN: + case WM8940_ADCBOOST: + case WM8940_OUTPUTCTL: + case WM8940_SPKMIX: + case WM8940_SPKVOL: + case WM8940_MONOMIX: + return true; + default: + return false; } } -static u16 wm8940_reg_defaults[] = { - 0x8940, /* Soft Reset */ - 0x0000, /* Power 1 */ - 0x0000, /* Power 2 */ - 0x0000, /* Power 3 */ - 0x0010, /* Interface Control */ - 0x0000, /* Companding Control */ - 0x0140, /* Clock Control */ - 0x0000, /* Additional Controls */ - 0x0000, /* GPIO Control */ - 0x0002, /* Auto Increment Control */ - 0x0000, /* DAC Control */ - 0x00FF, /* DAC Volume */ - 0, - 0, - 0x0100, /* ADC Control */ - 0x00FF, /* ADC Volume */ - 0x0000, /* Notch Filter 1 Control 1 */ - 0x0000, /* Notch Filter 1 Control 2 */ - 0x0000, /* Notch Filter 2 Control 1 */ - 0x0000, /* Notch Filter 2 Control 2 */ - 0x0000, /* Notch Filter 3 Control 1 */ - 0x0000, /* Notch Filter 3 Control 2 */ - 0x0000, /* Notch Filter 4 Control 1 */ - 0x0000, /* Notch Filter 4 Control 2 */ - 0x0032, /* DAC Limit Control 1 */ - 0x0000, /* DAC Limit Control 2 */ - 0, - 0, - 0, - 0, - 0, - 0, - 0x0038, /* ALC Control 1 */ - 0x000B, /* ALC Control 2 */ - 0x0032, /* ALC Control 3 */ - 0x0000, /* Noise Gate */ - 0x0041, /* PLLN */ - 0x000C, /* PLLK1 */ - 0x0093, /* PLLK2 */ - 0x00E9, /* PLLK3 */ - 0, - 0, - 0x0030, /* ALC Control 4 */ - 0, - 0x0002, /* Input Control */ - 0x0050, /* PGA Gain */ - 0, - 0x0002, /* ADC Boost Control */ - 0, - 0x0002, /* Output Control */ - 0x0000, /* Speaker Mixer Control */ - 0, - 0, - 0, - 0x0079, /* Speaker Volume */ - 0, - 0x0000, /* Mono Mixer Control */ +static const struct reg_default wm8940_reg_defaults[] = { + { 0x1, 0x0000 }, /* Power 1 */ + { 0x2, 0x0000 }, /* Power 2 */ + { 0x3, 0x0000 }, /* Power 3 */ + { 0x4, 0x0010 }, /* Interface Control */ + { 0x5, 0x0000 }, /* Companding Control */ + { 0x6, 0x0140 }, /* Clock Control */ + { 0x7, 0x0000 }, /* Additional Controls */ + { 0x8, 0x0000 }, /* GPIO Control */ + { 0x9, 0x0002 }, /* Auto Increment Control */ + { 0xa, 0x0000 }, /* DAC Control */ + { 0xb, 0x00FF }, /* DAC Volume */ + + { 0xe, 0x0100 }, /* ADC Control */ + { 0xf, 0x00FF }, /* ADC Volume */ + { 0x10, 0x0000 }, /* Notch Filter 1 Control 1 */ + { 0x11, 0x0000 }, /* Notch Filter 1 Control 2 */ + { 0x12, 0x0000 }, /* Notch Filter 2 Control 1 */ + { 0x13, 0x0000 }, /* Notch Filter 2 Control 2 */ + { 0x14, 0x0000 }, /* Notch Filter 3 Control 1 */ + { 0x15, 0x0000 }, /* Notch Filter 3 Control 2 */ + { 0x16, 0x0000 }, /* Notch Filter 4 Control 1 */ + { 0x17, 0x0000 }, /* Notch Filter 4 Control 2 */ + { 0x18, 0x0032 }, /* DAC Limit Control 1 */ + { 0x19, 0x0000 }, /* DAC Limit Control 2 */ + + { 0x20, 0x0038 }, /* ALC Control 1 */ + { 0x21, 0x000B }, /* ALC Control 2 */ + { 0x22, 0x0032 }, /* ALC Control 3 */ + { 0x23, 0x0000 }, /* Noise Gate */ + { 0x24, 0x0041 }, /* PLLN */ + { 0x25, 0x000C }, /* PLLK1 */ + { 0x26, 0x0093 }, /* PLLK2 */ + { 0x27, 0x00E9 }, /* PLLK3 */ + + { 0x2a, 0x0030 }, /* ALC Control 4 */ + + { 0x2c, 0x0002 }, /* Input Control */ + { 0x2d, 0x0050 }, /* PGA Gain */ + + { 0x2f, 0x0002 }, /* ADC Boost Control */ + + { 0x31, 0x0002 }, /* Output Control */ + { 0x32, 0x0000 }, /* Speaker Mixer Control */ + + { 0x36, 0x0079 }, /* Speaker Volume */ + + { 0x38, 0x0000 }, /* Mono Mixer Control */ }; static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; @@ -431,6 +469,7 @@ static int wm8940_mute(struct snd_soc_dai *dai, int mute) static int wm8940_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec); u16 val; u16 pwr_reg = snd_soc_read(codec, WM8940_POWER1) & 0x1F0; int ret = 0; @@ -454,7 +493,7 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8940->regmap); if (ret < 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); return ret; @@ -669,12 +708,11 @@ static int wm8940_resume(struct snd_soc_codec *codec) static int wm8940_probe(struct snd_soc_codec *codec) { - struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec); struct wm8940_setup_data *pdata = codec->dev->platform_data; int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8940->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -722,10 +760,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .num_dapm_widgets = ARRAY_SIZE(wm8940_dapm_widgets), .dapm_routes = wm8940_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8940_dapm_routes), - .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8940_reg_defaults, - .volatile_register = wm8940_volatile_register, +}; + +static const struct regmap_config wm8940_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8940_MONOMIX, + .reg_defaults = wm8940_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8940_reg_defaults), + + .readable_reg = wm8940_readable_register, + .volatile_reg = wm8940_volatile_register, }; static int wm8940_i2c_probe(struct i2c_client *i2c, @@ -739,8 +785,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, if (wm8940 == NULL) return -ENOMEM; + wm8940->regmap = devm_regmap_init_i2c(i2c, &wm8940_regmap); + if (IS_ERR(wm8940->regmap)) + return PTR_ERR(wm8940->regmap); + i2c_set_clientdata(i2c, wm8940); - wm8940->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8940, &wm8940_dai, 1); -- cgit v0.10.2 From c3e8494c001ce0bec0ebaa49c6f5eeb2aa5ab36a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:41 -0200 Subject: ASoC: wm8962: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 543c5c2..07da601 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -74,7 +74,7 @@ struct wm8962_priv { struct regulator_bulk_data supplies[WM8962_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8962_NUM_SUPPLIES]; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; @@ -3108,7 +3108,7 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 500, 1000, 2000, 4000, }; -- cgit v0.10.2 From e40e0b5da87bb4256a6dc62db7663b8a0c204f1f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Nov 2013 14:01:39 +0000 Subject: ASoC: wm8974: Convert to direct regmap API usage Moves us towards removing the ASoC level I/O functions. Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a2d01d1..15f45c7 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -27,22 +28,22 @@ #include "wm8974.h" -static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0050, 0x0000, 0x0140, 0x0000, - 0x0000, 0x0000, 0x0000, 0x00ff, - 0x0000, 0x0000, 0x0100, 0x00ff, - 0x0000, 0x0000, 0x012c, 0x002c, - 0x002c, 0x002c, 0x002c, 0x0000, - 0x0032, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0038, 0x000b, 0x0032, 0x0000, - 0x0008, 0x000c, 0x0093, 0x00e9, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0003, 0x0010, 0x0000, 0x0000, - 0x0000, 0x0002, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0039, 0x0000, - 0x0000, +static const struct reg_default wm8974_reg_defaults[] = { + { 0, 0x0000 }, { 1, 0x0000 }, { 2, 0x0000 }, { 3, 0x0000 }, + { 4, 0x0050 }, { 5, 0x0000 }, { 6, 0x0140 }, { 7, 0x0000 }, + { 8, 0x0000 }, { 9, 0x0000 }, { 10, 0x0000 }, { 11, 0x00ff }, + { 12, 0x0000 }, { 13, 0x0000 }, { 14, 0x0100 }, { 15, 0x00ff }, + { 16, 0x0000 }, { 17, 0x0000 }, { 18, 0x012c }, { 19, 0x002c }, + { 20, 0x002c }, { 21, 0x002c }, { 22, 0x002c }, { 23, 0x0000 }, + { 24, 0x0032 }, { 25, 0x0000 }, { 26, 0x0000 }, { 27, 0x0000 }, + { 28, 0x0000 }, { 29, 0x0000 }, { 30, 0x0000 }, { 31, 0x0000 }, + { 32, 0x0038 }, { 33, 0x000b }, { 34, 0x0032 }, { 35, 0x0000 }, + { 36, 0x0008 }, { 37, 0x000c }, { 38, 0x0093 }, { 39, 0x00e9 }, + { 40, 0x0000 }, { 41, 0x0000 }, { 42, 0x0000 }, { 43, 0x0000 }, + { 44, 0x0003 }, { 45, 0x0010 }, { 46, 0x0000 }, { 47, 0x0000 }, + { 48, 0x0000 }, { 49, 0x0002 }, { 50, 0x0000 }, { 51, 0x0000 }, + { 52, 0x0000 }, { 53, 0x0000 }, { 54, 0x0039 }, { 55, 0x0000 }, + { 56, 0x0000 }, }; #define WM8974_POWER1_BIASEN 0x08 @@ -514,7 +515,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(dev_get_regmap(codec->dev, NULL)); /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8974_POWER1, power1 | 0x3); @@ -579,11 +580,20 @@ static int wm8974_resume(struct snd_soc_codec *codec) return 0; } +static const struct regmap_config wm8974_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8974_MONOMIX, + .reg_defaults = wm8974_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults), +}; + static int wm8974_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -613,9 +623,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { .suspend = wm8974_suspend, .resume = wm8974_resume, .set_bias_level = wm8974_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8974_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8974_reg, .controls = wm8974_snd_controls, .num_controls = ARRAY_SIZE(wm8974_snd_controls), @@ -628,8 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { static int wm8974_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct regmap *regmap; int ret; + regmap = devm_regmap_init_i2c(i2c, &wm8974_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8974, &wm8974_dai, 1); -- cgit v0.10.2 From 6a0773368619644432fbe46f64ded6aa204baab9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 11:16:18 +0000 Subject: ASoC: wm8991: Use a supply to manage input power Instead of using a fake register and events to manage input power use a supply to do the job, saving code and preparing for regmap conversion of the driver. Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 3a39df7..5078fc8 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -374,30 +374,6 @@ static const struct snd_kcontrol_new wm8991_snd_controls[] = { /* * _DAPM_ Controls */ -static int inmixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS); - - if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) | - (1 << WM8991_AINLMUX_PWR_BIT))) - reg |= WM8991_AINL_ENA; - else - reg &= ~WM8991_AINL_ENA; - - if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) | - (1 << WM8991_AINRMUX_PWR_BIT))) - reg |= WM8991_AINR_ENA; - else - reg &= ~WM8991_AINR_ENA; - - snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); - return 0; -} - static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -655,6 +631,11 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RIN2"), SND_SOC_DAPM_INPUT("Internal ADC Source"), + SND_SOC_DAPM_SUPPLY("INL", WM8991_POWER_MANAGEMENT_2, + WM8991_AINL_ENA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("INR", WM8991_POWER_MANAGEMENT_2, + WM8991_AINR_ENA_BIT, 0, NULL, 0), + /* DACs */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2, WM8991_ADCL_ENA_BIT, 0), @@ -676,26 +657,22 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)), /* INMIXL */ - SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0, + SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8991_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8991_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8991_dapm_inmixl_controls)), /* AINLMUX */ - SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0, - &wm8991_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0, + &wm8991_dapm_ainlmux_controls), /* INMIXR */ - SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0, + SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8991_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8991_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8991_dapm_inmixr_controls)), /* AINRMUX */ - SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0, - &wm8991_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0, + &wm8991_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -797,6 +774,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right ADC", NULL, "Internal ADC Source"}, /* Input Side */ + {"INMIXL", NULL, "INL"}, + {"AINLMUX", NULL, "INL"}, + {"INMIXR", NULL, "INR"}, + {"AINRMUX", NULL, "INR"}, /* LIN12 PGA */ {"LIN12 PGA", "LIN1 Switch", "LIN1"}, {"LIN12 PGA", "LIN2 Switch", "LIN2"}, diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h index 07707d8..08ed383 100644 --- a/sound/soc/codecs/wm8991.h +++ b/sound/soc/codecs/wm8991.h @@ -76,7 +76,6 @@ #define WM8991_PLL1 0x3C #define WM8991_PLL2 0x3D #define WM8991_PLL3 0x3E -#define WM8991_INTDRIVBITS 0x3F #define WM8991_REGISTER_COUNT 60 #define WM8991_MAX_REGISTER 0x3F @@ -807,14 +806,6 @@ */ #define WM8991_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ -/* - * R63 (0x3F) - Internal Driver Bits - */ -#define WM8991_INMIXL_PWR_BIT 0 -#define WM8991_AINLMUX_PWR_BIT 1 -#define WM8991_INMIXR_PWR_BIT 2 -#define WM8991_AINRMUX_PWR_BIT 3 - #define WM8991_MCLK_DIV 0 #define WM8991_DACCLK_DIV 1 #define WM8991_ADCCLK_DIV 2 -- cgit v0.10.2 From 898249958281ebf52288eb0d3ed6797e1fc4b0bd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 12:28:06 +0000 Subject: ASoC: wm8991: Convert to table based control and widget init Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 5078fc8..86aa33f 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -764,7 +764,7 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8991_dapm_routes[] = { /* Make DACs turn on when playing even if not mixed into any outputs */ {"Internal DAC Sink", NULL, "Left DAC"}, {"Internal DAC Sink", NULL, "Right DAC"}, @@ -1278,13 +1278,6 @@ static int wm8991_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_codec_controls(codec, wm8991_snd_controls, - ARRAY_SIZE(wm8991_snd_controls)); - - snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, - ARRAY_SIZE(wm8991_dapm_widgets)); - snd_soc_dapm_add_routes(&codec->dapm, audio_map, - ARRAY_SIZE(audio_map)); return 0; } @@ -1333,6 +1326,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { .suspend = wm8991_suspend, .resume = wm8991_resume, .set_bias_level = wm8991_set_bias_level, + .controls = wm8991_snd_controls, + .num_controls = ARRAY_SIZE(wm8991_snd_controls), + .dapm_widgets = wm8991_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets), + .dapm_routes = wm8991_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes), .reg_cache_size = WM8991_MAX_REGISTER + 1, .reg_word_size = sizeof(u16), .reg_cache_default = wm8991_reg_defs -- cgit v0.10.2 From a86652e51a8776bc0fe811e32ec3118f03c7e3bb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:28:55 +0000 Subject: ASoC: wm8991: Convert to direct regmap API usage Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 86aa33f..7006f97 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -31,76 +32,85 @@ #include "wm8991.h" struct wm8991_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int pcmclk; }; -static const u16 wm8991_reg_defs[] = { - 0x8991, /* R0 - Reset */ - 0x0000, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x4050, /* R4 - Audio Interface (1) */ - 0x4000, /* R5 - Audio Interface (2) */ - 0x01C8, /* R6 - Clocking (1) */ - 0x0000, /* R7 - Clocking (2) */ - 0x0040, /* R8 - Audio Interface (3) */ - 0x0040, /* R9 - Audio Interface (4) */ - 0x0004, /* R10 - DAC CTRL */ - 0x00C0, /* R11 - Left DAC Digital Volume */ - 0x00C0, /* R12 - Right DAC Digital Volume */ - 0x0000, /* R13 - Digital Side Tone */ - 0x0100, /* R14 - ADC CTRL */ - 0x00C0, /* R15 - Left ADC Digital Volume */ - 0x00C0, /* R16 - Right ADC Digital Volume */ - 0x0000, /* R17 */ - 0x0000, /* R18 - GPIO CTRL 1 */ - 0x1000, /* R19 - GPIO1 & GPIO2 */ - 0x1010, /* R20 - GPIO3 & GPIO4 */ - 0x1010, /* R21 - GPIO5 & GPIO6 */ - 0x8000, /* R22 - GPIOCTRL 2 */ - 0x0800, /* R23 - GPIO_POL */ - 0x008B, /* R24 - Left Line Input 1&2 Volume */ - 0x008B, /* R25 - Left Line Input 3&4 Volume */ - 0x008B, /* R26 - Right Line Input 1&2 Volume */ - 0x008B, /* R27 - Right Line Input 3&4 Volume */ - 0x0000, /* R28 - Left Output Volume */ - 0x0000, /* R29 - Right Output Volume */ - 0x0066, /* R30 - Line Outputs Volume */ - 0x0022, /* R31 - Out3/4 Volume */ - 0x0079, /* R32 - Left OPGA Volume */ - 0x0079, /* R33 - Right OPGA Volume */ - 0x0003, /* R34 - Speaker Volume */ - 0x0003, /* R35 - ClassD1 */ - 0x0000, /* R36 */ - 0x0100, /* R37 - ClassD3 */ - 0x0000, /* R38 */ - 0x0000, /* R39 - Input Mixer1 */ - 0x0000, /* R40 - Input Mixer2 */ - 0x0000, /* R41 - Input Mixer3 */ - 0x0000, /* R42 - Input Mixer4 */ - 0x0000, /* R43 - Input Mixer5 */ - 0x0000, /* R44 - Input Mixer6 */ - 0x0000, /* R45 - Output Mixer1 */ - 0x0000, /* R46 - Output Mixer2 */ - 0x0000, /* R47 - Output Mixer3 */ - 0x0000, /* R48 - Output Mixer4 */ - 0x0000, /* R49 - Output Mixer5 */ - 0x0000, /* R50 - Output Mixer6 */ - 0x0180, /* R51 - Out3/4 Mixer */ - 0x0000, /* R52 - Line Mixer1 */ - 0x0000, /* R53 - Line Mixer2 */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 - Additional Control */ - 0x0000, /* R56 - AntiPOP1 */ - 0x0000, /* R57 - AntiPOP2 */ - 0x0000, /* R58 - MICBIAS */ - 0x0000, /* R59 */ - 0x0008, /* R60 - PLL1 */ - 0x0031, /* R61 - PLL2 */ - 0x0026, /* R62 - PLL3 */ +static const struct reg_default wm8991_reg_defaults[] = { + { 1, 0x0000 }, /* R1 - Power Management (1) */ + { 2, 0x6000 }, /* R2 - Power Management (2) */ + { 3, 0x0000 }, /* R3 - Power Management (3) */ + { 4, 0x4050 }, /* R4 - Audio Interface (1) */ + { 5, 0x4000 }, /* R5 - Audio Interface (2) */ + { 6, 0x01C8 }, /* R6 - Clocking (1) */ + { 7, 0x0000 }, /* R7 - Clocking (2) */ + { 8, 0x0040 }, /* R8 - Audio Interface (3) */ + { 9, 0x0040 }, /* R9 - Audio Interface (4) */ + { 10, 0x0004 }, /* R10 - DAC CTRL */ + { 11, 0x00C0 }, /* R11 - Left DAC Digital Volume */ + { 12, 0x00C0 }, /* R12 - Right DAC Digital Volume */ + { 13, 0x0000 }, /* R13 - Digital Side Tone */ + { 14, 0x0100 }, /* R14 - ADC CTRL */ + { 15, 0x00C0 }, /* R15 - Left ADC Digital Volume */ + { 16, 0x00C0 }, /* R16 - Right ADC Digital Volume */ + + { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ + { 19, 0x1000 }, /* R19 - GPIO1 & GPIO2 */ + { 20, 0x1010 }, /* R20 - GPIO3 & GPIO4 */ + { 21, 0x1010 }, /* R21 - GPIO5 & GPIO6 */ + { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 23, 0x0800 }, /* R23 - GPIO_POL */ + { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ + { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ + { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ + { 27, 0x008B }, /* R27 - Right Line Input 3&4 Volume */ + { 28, 0x0000 }, /* R28 - Left Output Volume */ + { 29, 0x0000 }, /* R29 - Right Output Volume */ + { 30, 0x0066 }, /* R30 - Line Outputs Volume */ + { 31, 0x0022 }, /* R31 - Out3/4 Volume */ + { 32, 0x0079 }, /* R32 - Left OPGA Volume */ + { 33, 0x0079 }, /* R33 - Right OPGA Volume */ + { 34, 0x0003 }, /* R34 - Speaker Volume */ + { 35, 0x0003 }, /* R35 - ClassD1 */ + + { 37, 0x0100 }, /* R37 - ClassD3 */ + + { 39, 0x0000 }, /* R39 - Input Mixer1 */ + { 40, 0x0000 }, /* R40 - Input Mixer2 */ + { 41, 0x0000 }, /* R41 - Input Mixer3 */ + { 42, 0x0000 }, /* R42 - Input Mixer4 */ + { 43, 0x0000 }, /* R43 - Input Mixer5 */ + { 44, 0x0000 }, /* R44 - Input Mixer6 */ + { 45, 0x0000 }, /* R45 - Output Mixer1 */ + { 46, 0x0000 }, /* R46 - Output Mixer2 */ + { 47, 0x0000 }, /* R47 - Output Mixer3 */ + { 48, 0x0000 }, /* R48 - Output Mixer4 */ + { 49, 0x0000 }, /* R49 - Output Mixer5 */ + { 50, 0x0000 }, /* R50 - Output Mixer6 */ + { 51, 0x0180 }, /* R51 - Out3/4 Mixer */ + { 52, 0x0000 }, /* R52 - Line Mixer1 */ + { 53, 0x0000 }, /* R53 - Line Mixer2 */ + { 54, 0x0000 }, /* R54 - Speaker Mixer */ + { 55, 0x0000 }, /* R55 - Additional Control */ + { 56, 0x0000 }, /* R56 - AntiPOP1 */ + { 57, 0x0000 }, /* R57 - AntiPOP2 */ + { 58, 0x0000 }, /* R58 - MICBIAS */ + + { 60, 0x0008 }, /* R60 - PLL1 */ + { 61, 0x0031 }, /* R61 - PLL2 */ + { 62, 0x0026 }, /* R62 - PLL3 */ }; +static bool wm8991_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8991_RESET: + return true; + default: + return false; + } +} + #define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) static const unsigned int rec_mix_tlv[] = { @@ -1110,6 +1120,7 @@ static int wm8991_mute(struct snd_soc_dai *dai, int mute) static int wm8991_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8991_priv *wm8991 = snd_soc_codec_get_drvdata(codec); u16 val; switch (level) { @@ -1125,7 +1136,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8991->regmap); /* Enable all output discharge bits */ snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | WM8991_DIS_RLINE | WM8991_DIS_OUT3 | @@ -1213,7 +1224,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8991_ANTIPOP2, 0x0); - codec->cache_sync = 1; + regcache_mark_dirty(wm8991->regmap); break; } @@ -1247,7 +1258,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) wm8991 = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -1332,9 +1343,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { .num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets), .dapm_routes = wm8991_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes), - .reg_cache_size = WM8991_MAX_REGISTER + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8991_reg_defs +}; + +static const struct regmap_config wm8991_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8991_PLL3, + .volatile_reg = wm8991_volatile, + .reg_defaults = wm8991_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8991_reg_defaults), + .cache_type = REGCACHE_RBTREE, }; static int wm8991_i2c_probe(struct i2c_client *i2c, @@ -1347,7 +1366,10 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, if (!wm8991) return -ENOMEM; - wm8991->control_type = SND_SOC_I2C; + wm8991->regmap = devm_regmap_init_i2c(i2c, &wm8991_regmap); + if (IS_ERR(wm8991->regmap)) + return PTR_ERR(wm8991->regmap); + i2c_set_clientdata(i2c, wm8991); ret = snd_soc_register_codec(&i2c->dev, -- cgit v0.10.2 From e4634804cacce6fe1ec34d92786f764fcb75cb97 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:32:36 +0000 Subject: ASoC: wm8991: Move basic initialisation to I2C level probe This is better practice, though some of this stuff ought not to be here at all. Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 7006f97..5fdcf78 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -111,8 +111,6 @@ static bool wm8991_volatile(struct device *dev, unsigned int reg) } } -#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) - static const unsigned int rec_mix_tlv[] = { TLV_DB_RANGE_HEAD(1), 0, 7, TLV_DB_LINEAR_ITEM(-1500, 600), @@ -1264,31 +1262,8 @@ static int wm8991_probe(struct snd_soc_codec *codec) return ret; } - ret = wm8991_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_update_bits(codec, WM8991_AUDIO_INTERFACE_4, - WM8991_ALRCGPIO1, WM8991_ALRCGPIO1); - - snd_soc_update_bits(codec, WM8991_GPIO1_GPIO2, - WM8991_GPIO1_SEL_MASK, 1); - - snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_1, - WM8991_VREF_ENA | WM8991_VMID_MODE_MASK, - WM8991_VREF_ENA | WM8991_VMID_MODE_MASK); - - snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_2, - WM8991_OPCLK_ENA, WM8991_OPCLK_ENA); - - snd_soc_write(codec, WM8991_DAC_CTRL, 0); - snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - return 0; } @@ -1372,6 +1347,31 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8991); + ret = regmap_write(wm8991->regmap, WM8991_RESET, 0); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + regmap_update_bits(wm8991->regmap, WM8991_AUDIO_INTERFACE_4, + WM8991_ALRCGPIO1, WM8991_ALRCGPIO1); + + regmap_update_bits(wm8991->regmap, WM8991_GPIO1_GPIO2, + WM8991_GPIO1_SEL_MASK, 1); + + regmap_update_bits(wm8991->regmap, WM8991_POWER_MANAGEMENT_1, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK); + + regmap_update_bits(wm8991->regmap, WM8991_POWER_MANAGEMENT_2, + WM8991_OPCLK_ENA, WM8991_OPCLK_ENA); + + regmap_write(wm8991->regmap, WM8991_DAC_CTRL, 0); + regmap_write(wm8991->regmap, WM8991_LEFT_OUTPUT_VOLUME, + 0x50 | (1<<8)); + regmap_write(wm8991->regmap, WM8991_RIGHT_OUTPUT_VOLUME, + 0x50 | (1<<8)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8991, &wm8991_dai, 1); -- cgit v0.10.2 From a0a05916cf67a007f4ee0071fd0fa04e45137a38 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:34:48 +0000 Subject: ASoC: wm8991: Verify device ID during probe() Just in case. Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 5fdcf78..dba0306 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1335,6 +1335,7 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8991_priv *wm8991; + unsigned int val; int ret; wm8991 = devm_kzalloc(&i2c->dev, sizeof(*wm8991), GFP_KERNEL); @@ -1347,6 +1348,16 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8991); + ret = regmap_read(wm8991->regmap, WM8991_RESET, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read device ID: %d\n", ret); + return ret; + } + if (val != 0x8991) { + dev_err(&i2c->dev, "Device with ID %x is not a WM8991\n", val); + return -EINVAL; + } + ret = regmap_write(wm8991->regmap, WM8991_RESET, 0); if (ret < 0) { dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); -- cgit v0.10.2 From 4f534777c130180f9338f0fb96090d43464b7ddf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 15:14:48 +0000 Subject: ASoC: ak4641: Convert to direct regmap API usage We're trying to remove the ASoC level I/O functions. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 49cc5f6..94cbe50 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -30,6 +31,7 @@ /* codec private data */ struct ak4641_priv { + struct regmap *regmap; unsigned int sysclk; int deemph; int playback_fs; @@ -38,12 +40,12 @@ struct ak4641_priv { /* * ak4641 register cache */ -static const u8 ak4641_reg[AK4641_CACHEREGNUM] = { - 0x00, 0x80, 0x00, 0x80, - 0x02, 0x00, 0x11, 0x05, - 0x00, 0x00, 0x36, 0x10, - 0x00, 0x00, 0x57, 0x00, - 0x88, 0x88, 0x08, 0x08 +static const struct reg_default ak4641_reg_defaults[] = { + { 0, 0x00 }, { 1, 0x80 }, { 2, 0x00 }, { 3, 0x80 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x11 }, { 7, 0x05 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x36 }, { 11, 0x10 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x57 }, { 15, 0x00 }, + { 16, 0x88 }, { 17, 0x88 }, { 18, 0x08 }, { 19, 0x08 } }; static const int deemph_settings[] = {44100, 0, 48000, 32000}; @@ -396,6 +398,7 @@ static int ak4641_mute(struct snd_soc_dai *dai, int mute) static int ak4641_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; @@ -417,7 +420,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 1); mdelay(1); - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(ak4641->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -433,7 +436,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 0); if (pdata && gpio_is_valid(pdata->gpio_power)) gpio_set_value(pdata->gpio_power, 0); - codec->cache_sync = 1; + regcache_mark_dirty(ak4641->regmap); break; } codec->dapm.bias_level = level; @@ -518,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -550,12 +553,17 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { .dapm_routes = ak4641_audio_map, .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), .set_bias_level = ak4641_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ak4641_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4641_reg, - .reg_cache_step = 1, }; +static const struct regmap_config ak4641_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4641_BTIF, + .reg_defaults = ak4641_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4641_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; static int ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -569,6 +577,10 @@ static int ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + ak4641->regmap = devm_regmap_init_i2c(i2c, &ak4641_regmap); + if (IS_ERR(ak4641->regmap)) + return PTR_ERR(ak4641->regmap); + if (pdata) { if (gpio_is_valid(pdata->gpio_power)) { ret = gpio_request_one(pdata->gpio_power, -- cgit v0.10.2 From 9a3e1b8c503590a2a89164586c0e65415bb90979 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 19 Nov 2013 16:04:01 +0000 Subject: ASoC: wm5110: Hook up ADSP2 cores Signed-off-by: Mark Brown Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b33b45d..983d087a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -163,8 +163,10 @@ config SND_SOC_WM_HUBS config SND_SOC_WM_ADSP tristate default y if SND_SOC_WM5102=y + default y if SND_SOC_WM5110=y default y if SND_SOC_WM2200=y default m if SND_SOC_WM5102=m + default m if SND_SOC_WM5110=m default m if SND_SOC_WM2200=m config SND_SOC_AB8500_CODEC diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c6dbc1d..e5f2804 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -30,13 +30,51 @@ #include #include "arizona.h" +#include "wm_adsp.h" #include "wm5110.h" +#define WM5110_NUM_ADSP 4 + struct wm5110_priv { struct arizona_priv core; struct arizona_fll fll[2]; }; +static const struct wm_adsp_region wm5110_dsp1_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x100000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x180000 }, + { .type = WMFW_ADSP2_XM, .base = 0x190000 }, + { .type = WMFW_ADSP2_YM, .base = 0x1a8000 }, +}; + +static const struct wm_adsp_region wm5110_dsp2_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x200000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x280000 }, + { .type = WMFW_ADSP2_XM, .base = 0x290000 }, + { .type = WMFW_ADSP2_YM, .base = 0x2a8000 }, +}; + +static const struct wm_adsp_region wm5110_dsp3_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x300000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x380000 }, + { .type = WMFW_ADSP2_XM, .base = 0x390000 }, + { .type = WMFW_ADSP2_YM, .base = 0x3a8000 }, +}; + +static const struct wm_adsp_region wm5110_dsp4_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x400000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x480000 }, + { .type = WMFW_ADSP2_XM, .base = 0x490000 }, + { .type = WMFW_ADSP2_YM, .base = 0x4a8000 }, +}; + +static const struct wm_adsp_region *wm5110_dsp_regions[] = { + wm5110_dsp1_regions, + wm5110_dsp2_regions, + wm5110_dsp3_regions, + wm5110_dsp4_regions, +}; + static const struct reg_default wm5110_sysclk_revd_patch[] = { { 0x3093, 0x1001 }, { 0x30E3, 0x1301 }, @@ -395,6 +433,22 @@ ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP1L, ARIZONA_DSP1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP1R, ARIZONA_DSP1RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP1, ARIZONA_DSP1AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP2L, ARIZONA_DSP2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP2R, ARIZONA_DSP2RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP2, ARIZONA_DSP2AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP3L, ARIZONA_DSP3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP3R, ARIZONA_DSP3RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP3, ARIZONA_DSP3AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP4L, ARIZONA_DSP4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP4R, ARIZONA_DSP4RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP4, ARIZONA_DSP4AUX1MIX_INPUT_1_SOURCE); + ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); @@ -587,6 +641,11 @@ SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), +WM_ADSP2("DSP1", 0), +WM_ADSP2("DSP2", 1), +WM_ADSP2("DSP3", 2), +WM_ADSP2("DSP4", 3), + SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5110_aec_loopback_mux), @@ -809,6 +868,11 @@ ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"), +ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), +ARIZONA_DSP_WIDGETS(DSP2, "DSP2"), +ARIZONA_DSP_WIDGETS(DSP3, "DSP3"), +ARIZONA_DSP_WIDGETS(DSP4, "DSP4"), + SND_SOC_DAPM_OUTPUT("HPOUT1L"), SND_SOC_DAPM_OUTPUT("HPOUT1R"), SND_SOC_DAPM_OUTPUT("HPOUT2L"), @@ -881,7 +945,31 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "ASRC1L", "ASRC1L" }, \ { name, "ASRC1R", "ASRC1R" }, \ { name, "ASRC2L", "ASRC2L" }, \ - { name, "ASRC2R", "ASRC2R" } + { name, "ASRC2R", "ASRC2R" }, \ + { name, "DSP1.1", "DSP1" }, \ + { name, "DSP1.2", "DSP1" }, \ + { name, "DSP1.3", "DSP1" }, \ + { name, "DSP1.4", "DSP1" }, \ + { name, "DSP1.5", "DSP1" }, \ + { name, "DSP1.6", "DSP1" }, \ + { name, "DSP2.1", "DSP2" }, \ + { name, "DSP2.2", "DSP2" }, \ + { name, "DSP2.3", "DSP2" }, \ + { name, "DSP2.4", "DSP2" }, \ + { name, "DSP2.5", "DSP2" }, \ + { name, "DSP2.6", "DSP2" }, \ + { name, "DSP3.1", "DSP3" }, \ + { name, "DSP3.2", "DSP3" }, \ + { name, "DSP3.3", "DSP3" }, \ + { name, "DSP3.4", "DSP3" }, \ + { name, "DSP3.5", "DSP3" }, \ + { name, "DSP3.6", "DSP3" }, \ + { name, "DSP4.1", "DSP4" }, \ + { name, "DSP4.2", "DSP4" }, \ + { name, "DSP4.3", "DSP4" }, \ + { name, "DSP4.4", "DSP4" }, \ + { name, "DSP4.5", "DSP4" }, \ + { name, "DSP4.6", "DSP4" } static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "DBVDD2" }, @@ -1087,6 +1175,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + ARIZONA_DSP_ROUTES("DSP1"), + ARIZONA_DSP_ROUTES("DSP2"), + ARIZONA_DSP_ROUTES("DSP3"), + ARIZONA_DSP_ROUTES("DSP4"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, @@ -1292,6 +1385,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); + if (ret != 0) + return ret; + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; @@ -1346,7 +1443,7 @@ static int wm5110_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); struct wm5110_priv *wm5110; - int i; + int i, ret; wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv), GFP_KERNEL); @@ -1357,6 +1454,24 @@ static int wm5110_probe(struct platform_device *pdev) wm5110->core.arizona = arizona; wm5110->core.num_inputs = 8; + for (i = 0; i < WM5110_NUM_ADSP; i++) { + wm5110->core.adsp[i].part = "wm5110"; + wm5110->core.adsp[i].num = i + 1; + wm5110->core.adsp[i].type = WMFW_ADSP2; + wm5110->core.adsp[i].dev = arizona->dev; + wm5110->core.adsp[i].regmap = arizona->regmap; + + wm5110->core.adsp[i].base = ARIZONA_DSP1_CONTROL_1 + + (0x100 * i); + wm5110->core.adsp[i].mem = wm5110_dsp_regions[i]; + wm5110->core.adsp[i].num_mems + = ARRAY_SIZE(wm5110_dsp1_regions); + + ret = wm_adsp2_init(&wm5110->core.adsp[i], false); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) wm5110->fll[i].vco_mult = 3; -- cgit v0.10.2 From 0da2e5baf4233e2744a7fc691932638c39d9b245 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Nov 2013 16:04:02 +0000 Subject: ASoC: wm5110: Add basic support for ISRCs Add support for the ISRCs that matches the current support on the w5102. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index e5f2804..0e63d8c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -277,6 +277,10 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), + ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), @@ -501,6 +505,36 @@ ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT3, ARIZONA_ISRC2INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT4, ARIZONA_ISRC2INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC3, ARIZONA_ISRC2DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC4, ARIZONA_ISRC2DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC3INT1, ARIZONA_ISRC3INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT2, ARIZONA_ISRC3INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT3, ARIZONA_ISRC3INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT4, ARIZONA_ISRC3INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC3DEC1, ARIZONA_ISRC3DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC2, ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC3, ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC4, ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE); + static const char *wm5110_aec_loopback_texts[] = { "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "HPOUT3L", "HPOUT3R", "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", "SPKDAT2L", "SPKDAT2R", @@ -646,6 +680,60 @@ WM_ADSP2("DSP2", 1), WM_ADSP2("DSP3", 2), WM_ADSP2("DSP4", 3), +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT3", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT4", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC3", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC4", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC3INT1", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT2", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT3", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT4", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC3DEC1", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC2", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC3", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5110_aec_loopback_mux), @@ -873,6 +961,36 @@ ARIZONA_DSP_WIDGETS(DSP2, "DSP2"), ARIZONA_DSP_WIDGETS(DSP3, "DSP3"), ARIZONA_DSP_WIDGETS(DSP4, "DSP4"), +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), +ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"), +ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), +ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"), +ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), +ARIZONA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"), +ARIZONA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), +ARIZONA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"), +ARIZONA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"), + +ARIZONA_MUX_WIDGETS(ISRC3DEC1, "ISRC3DEC1"), +ARIZONA_MUX_WIDGETS(ISRC3DEC2, "ISRC3DEC2"), +ARIZONA_MUX_WIDGETS(ISRC3DEC3, "ISRC3DEC3"), +ARIZONA_MUX_WIDGETS(ISRC3DEC4, "ISRC3DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC3INT1, "ISRC3INT1"), +ARIZONA_MUX_WIDGETS(ISRC3INT2, "ISRC3INT2"), +ARIZONA_MUX_WIDGETS(ISRC3INT3, "ISRC3INT3"), +ARIZONA_MUX_WIDGETS(ISRC3INT4, "ISRC3INT4"), + SND_SOC_DAPM_OUTPUT("HPOUT1L"), SND_SOC_DAPM_OUTPUT("HPOUT1R"), SND_SOC_DAPM_OUTPUT("HPOUT2L"), @@ -946,6 +1064,30 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "ASRC1R", "ASRC1R" }, \ { name, "ASRC2L", "ASRC2L" }, \ { name, "ASRC2R", "ASRC2R" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1DEC3", "ISRC1DEC3" }, \ + { name, "ISRC1DEC4", "ISRC1DEC4" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC1INT3", "ISRC1INT3" }, \ + { name, "ISRC1INT4", "ISRC1INT4" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2DEC3", "ISRC2DEC3" }, \ + { name, "ISRC2DEC4", "ISRC2DEC4" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" }, \ + { name, "ISRC2INT3", "ISRC2INT3" }, \ + { name, "ISRC2INT4", "ISRC2INT4" }, \ + { name, "ISRC3DEC1", "ISRC3DEC1" }, \ + { name, "ISRC3DEC2", "ISRC3DEC2" }, \ + { name, "ISRC3DEC3", "ISRC3DEC3" }, \ + { name, "ISRC3DEC4", "ISRC3DEC4" }, \ + { name, "ISRC3INT1", "ISRC3INT1" }, \ + { name, "ISRC3INT2", "ISRC3INT2" }, \ + { name, "ISRC3INT3", "ISRC3INT3" }, \ + { name, "ISRC3INT4", "ISRC3INT4" }, \ { name, "DSP1.1", "DSP1" }, \ { name, "DSP1.2", "DSP1" }, \ { name, "DSP1.3", "DSP1" }, \ @@ -1180,6 +1322,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_DSP_ROUTES("DSP3"), ARIZONA_DSP_ROUTES("DSP4"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"), + ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"), + ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"), + ARIZONA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"), + ARIZONA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"), + + ARIZONA_MUX_ROUTES("ISRC3INT1", "ISRC3INT1"), + ARIZONA_MUX_ROUTES("ISRC3INT2", "ISRC3INT2"), + ARIZONA_MUX_ROUTES("ISRC3INT3", "ISRC3INT3"), + ARIZONA_MUX_ROUTES("ISRC3INT4", "ISRC3INT4"), + + ARIZONA_MUX_ROUTES("ISRC3DEC1", "ISRC3DEC1"), + ARIZONA_MUX_ROUTES("ISRC3DEC2", "ISRC3DEC2"), + ARIZONA_MUX_ROUTES("ISRC3DEC3", "ISRC3DEC3"), + ARIZONA_MUX_ROUTES("ISRC3DEC4", "ISRC3DEC4"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, @@ -1482,6 +1654,12 @@ static int wm5110_probe(struct platform_device *pdev) ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, &wm5110->fll[1]); + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++) arizona_init_dai(&wm5110->core, i); -- cgit v0.10.2 From 1b49f10c69b45a8b169c69eaa127e7f91a32f7c7 Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 16 Oct 2013 21:10:52 +0200 Subject: spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX With support for amba-pl08x driver, on S3C64xx the generic DMA engine API can be used instead of the private s3c-dma interface. Signed-off-by: Tomasz Figa Signed-off-by: Mark Brown diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index eb1f1ef..760d7b6 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -395,7 +395,7 @@ config SPI_S3C24XX_FIQ config SPI_S3C64XX tristate "Samsung S3C64XX series type SPI" depends on PLAT_SAMSUNG - select S3C64XX_DMA if ARCH_S3C64XX + select S3C64XX_DMA if ARCH_S3C64XX && !S3C64XX_PL080 help SPI driver for Samsung S3C64XX and newer SoCs. -- cgit v0.10.2 From 1db0287ab186637488e51ae43109692624f2d5a3 Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 16 Oct 2013 21:10:54 +0200 Subject: ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver This patch adds all required platform-specific data and initialization code to support the generic amba-pl08x driver on S3C64xx SoCs. Also some compatibility definitions are added to make the transition from legacy API to DMA engine easier. The biggest hack here is passing const char * pointers through DMA resource, casted to unsigned long, but this is how Samsung DMA wrappers (used to support both s3c-dma and DMA engine in drivers) are designed. Signed-off-by: Tomasz Figa Signed-off-by: Mark Brown diff --git a/arch/arm/Kconfig b/arch/arm/Kconfig index c1f1a7e..ba0e232 100644 --- a/arch/arm/Kconfig +++ b/arch/arm/Kconfig @@ -723,6 +723,7 @@ config ARCH_S3C64XX bool "Samsung S3C64XX" select ARCH_HAS_CPUFREQ select ARCH_REQUIRE_GPIOLIB + select ARM_AMBA select ARM_VIC select CLKDEV_LOOKUP select CLKSRC_SAMSUNG_PWM diff --git a/arch/arm/mach-s3c64xx/Kconfig b/arch/arm/mach-s3c64xx/Kconfig index 2cb8dc5..d8e0288 100644 --- a/arch/arm/mach-s3c64xx/Kconfig +++ b/arch/arm/mach-s3c64xx/Kconfig @@ -17,9 +17,15 @@ config CPU_S3C6410 help Enable S3C6410 CPU support +config S3C64XX_PL080 + bool "S3C64XX DMA using generic PL08x driver" + select AMBA_PL08X + select SAMSUNG_DMADEV + config S3C64XX_DMA - bool "S3C64XX DMA" + bool "S3C64XX DMA using legacy S3C DMA API" select S3C_DMA + depends on !S3C64XX_PL080 config S3C64XX_SETUP_SDHCI bool diff --git a/arch/arm/mach-s3c64xx/Makefile b/arch/arm/mach-s3c64xx/Makefile index 6faedcf..e8e9a46 100644 --- a/arch/arm/mach-s3c64xx/Makefile +++ b/arch/arm/mach-s3c64xx/Makefile @@ -27,6 +27,7 @@ obj-$(CONFIG_CPU_IDLE) += cpuidle.o # DMA support obj-$(CONFIG_S3C64XX_DMA) += dma.o +obj-$(CONFIG_S3C64XX_PL080) += pl080.o # Device support diff --git a/arch/arm/mach-s3c64xx/common.h b/arch/arm/mach-s3c64xx/common.h index bd3bd56..7043e7a 100644 --- a/arch/arm/mach-s3c64xx/common.h +++ b/arch/arm/mach-s3c64xx/common.h @@ -58,4 +58,9 @@ int __init s3c64xx_pm_late_initcall(void); static inline int s3c64xx_pm_late_initcall(void) { return 0; } #endif +#ifdef CONFIG_S3C64XX_PL080 +extern struct pl08x_platform_data s3c64xx_dma0_plat_data; +extern struct pl08x_platform_data s3c64xx_dma1_plat_data; +#endif + #endif /* __ARCH_ARM_MACH_S3C64XX_COMMON_H */ diff --git a/arch/arm/mach-s3c64xx/include/mach/dma.h b/arch/arm/mach-s3c64xx/include/mach/dma.h index fe1a98c..26a6bc3 100644 --- a/arch/arm/mach-s3c64xx/include/mach/dma.h +++ b/arch/arm/mach-s3c64xx/include/mach/dma.h @@ -11,6 +11,8 @@ #ifndef __ASM_ARCH_DMA_H #define __ASM_ARCH_DMA_H __FILE__ +#ifdef CONFIG_S3C64XX_DMA + #define S3C_DMA_CHANNELS (16) /* see mach-s3c2410/dma.h for notes on dma channel numbers */ @@ -128,4 +130,65 @@ struct s3c2410_dma_chan { #include +#else + +#define S3C64XX_DMA_CHAN(name) ((unsigned long)(name)) + +/* DMA0/SDMA0 */ +#define DMACH_UART0 S3C64XX_DMA_CHAN("uart0_tx") +#define DMACH_UART0_SRC2 S3C64XX_DMA_CHAN("uart0_rx") +#define DMACH_UART1 S3C64XX_DMA_CHAN("uart1_tx") +#define DMACH_UART1_SRC2 S3C64XX_DMA_CHAN("uart1_rx") +#define DMACH_UART2 S3C64XX_DMA_CHAN("uart2_tx") +#define DMACH_UART2_SRC2 S3C64XX_DMA_CHAN("uart2_rx") +#define DMACH_UART3 S3C64XX_DMA_CHAN("uart3_tx") +#define DMACH_UART3_SRC2 S3C64XX_DMA_CHAN("uart3_rx") +#define DMACH_PCM0_TX S3C64XX_DMA_CHAN("pcm0_tx") +#define DMACH_PCM0_RX S3C64XX_DMA_CHAN("pcm0_rx") +#define DMACH_I2S0_OUT S3C64XX_DMA_CHAN("i2s0_tx") +#define DMACH_I2S0_IN S3C64XX_DMA_CHAN("i2s0_rx") +#define DMACH_SPI0_TX S3C64XX_DMA_CHAN("spi0_tx") +#define DMACH_SPI0_RX S3C64XX_DMA_CHAN("spi0_rx") +#define DMACH_HSI_I2SV40_TX S3C64XX_DMA_CHAN("i2s2_tx") +#define DMACH_HSI_I2SV40_RX S3C64XX_DMA_CHAN("i2s2_rx") + +/* DMA1/SDMA1 */ +#define DMACH_PCM1_TX S3C64XX_DMA_CHAN("pcm1_tx") +#define DMACH_PCM1_RX S3C64XX_DMA_CHAN("pcm1_rx") +#define DMACH_I2S1_OUT S3C64XX_DMA_CHAN("i2s1_tx") +#define DMACH_I2S1_IN S3C64XX_DMA_CHAN("i2s1_rx") +#define DMACH_SPI1_TX S3C64XX_DMA_CHAN("spi1_tx") +#define DMACH_SPI1_RX S3C64XX_DMA_CHAN("spi1_rx") +#define DMACH_AC97_PCMOUT S3C64XX_DMA_CHAN("ac97_out") +#define DMACH_AC97_PCMIN S3C64XX_DMA_CHAN("ac97_in") +#define DMACH_AC97_MICIN S3C64XX_DMA_CHAN("ac97_mic") +#define DMACH_PWM S3C64XX_DMA_CHAN("pwm") +#define DMACH_IRDA S3C64XX_DMA_CHAN("irda") +#define DMACH_EXTERNAL S3C64XX_DMA_CHAN("external") +#define DMACH_SECURITY_RX S3C64XX_DMA_CHAN("sec_rx") +#define DMACH_SECURITY_TX S3C64XX_DMA_CHAN("sec_tx") + +enum dma_ch { + DMACH_MAX = 32 +}; + +struct s3c2410_dma_client { + char *name; +}; + +static inline bool samsung_dma_has_circular(void) +{ + return true; +} + +static inline bool samsung_dma_is_dmadev(void) +{ + return true; +} + +#include +#include + +#endif + #endif /* __ASM_ARCH_IRQ_H */ diff --git a/arch/arm/mach-s3c64xx/pl080.c b/arch/arm/mach-s3c64xx/pl080.c new file mode 100644 index 0000000..901a984 --- /dev/null +++ b/arch/arm/mach-s3c64xx/pl080.c @@ -0,0 +1,244 @@ +/* + * Samsung's S3C64XX generic DMA support using amba-pl08x driver. + * + * Copyright (c) 2013 Tomasz Figa + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include + +#include +#include + +#include "regs-sys.h" + +static int pl08x_get_xfer_signal(const struct pl08x_channel_data *cd) +{ + return cd->min_signal; +} + +static void pl08x_put_xfer_signal(const struct pl08x_channel_data *cd, int ch) +{ +} + +/* + * DMA0 + */ + +static struct pl08x_channel_data s3c64xx_dma0_info[] = { + { + .bus_id = "uart0_tx", + .min_signal = 0, + .max_signal = 0, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart0_rx", + .min_signal = 1, + .max_signal = 1, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart1_tx", + .min_signal = 2, + .max_signal = 2, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart1_rx", + .min_signal = 3, + .max_signal = 3, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart2_tx", + .min_signal = 4, + .max_signal = 4, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart2_rx", + .min_signal = 5, + .max_signal = 5, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart3_tx", + .min_signal = 6, + .max_signal = 6, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "uart3_rx", + .min_signal = 7, + .max_signal = 7, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "pcm0_tx", + .min_signal = 8, + .max_signal = 8, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "pcm0_rx", + .min_signal = 9, + .max_signal = 9, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "i2s0_tx", + .min_signal = 10, + .max_signal = 10, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "i2s0_rx", + .min_signal = 11, + .max_signal = 11, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "spi0_tx", + .min_signal = 12, + .max_signal = 12, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "spi0_rx", + .min_signal = 13, + .max_signal = 13, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "i2s2_tx", + .min_signal = 14, + .max_signal = 14, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "i2s2_rx", + .min_signal = 15, + .max_signal = 15, + .periph_buses = PL08X_AHB2, + } +}; + +struct pl08x_platform_data s3c64xx_dma0_plat_data = { + .memcpy_channel = { + .bus_id = "memcpy", + .cctl_memcpy = + (PL080_BSIZE_4 << PL080_CONTROL_SB_SIZE_SHIFT | + PL080_BSIZE_4 << PL080_CONTROL_DB_SIZE_SHIFT | + PL080_WIDTH_32BIT << PL080_CONTROL_SWIDTH_SHIFT | + PL080_WIDTH_32BIT << PL080_CONTROL_DWIDTH_SHIFT | + PL080_CONTROL_PROT_BUFF | PL080_CONTROL_PROT_CACHE | + PL080_CONTROL_PROT_SYS), + }, + .lli_buses = PL08X_AHB1, + .mem_buses = PL08X_AHB1, + .get_xfer_signal = pl08x_get_xfer_signal, + .put_xfer_signal = pl08x_put_xfer_signal, + .slave_channels = s3c64xx_dma0_info, + .num_slave_channels = ARRAY_SIZE(s3c64xx_dma0_info), +}; + +static AMBA_AHB_DEVICE(s3c64xx_dma0, "dma-pl080s.0", 0, + 0x75000000, {IRQ_DMA0}, &s3c64xx_dma0_plat_data); + +/* + * DMA1 + */ + +static struct pl08x_channel_data s3c64xx_dma1_info[] = { + { + .bus_id = "pcm1_tx", + .min_signal = 0, + .max_signal = 0, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "pcm1_rx", + .min_signal = 1, + .max_signal = 1, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "i2s1_tx", + .min_signal = 2, + .max_signal = 2, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "i2s1_rx", + .min_signal = 3, + .max_signal = 3, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "spi1_tx", + .min_signal = 4, + .max_signal = 4, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "spi1_rx", + .min_signal = 5, + .max_signal = 5, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "ac97_out", + .min_signal = 6, + .max_signal = 6, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "ac97_in", + .min_signal = 7, + .max_signal = 7, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "ac97_mic", + .min_signal = 8, + .max_signal = 8, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "pwm", + .min_signal = 9, + .max_signal = 9, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "irda", + .min_signal = 10, + .max_signal = 10, + .periph_buses = PL08X_AHB2, + }, { + .bus_id = "external", + .min_signal = 11, + .max_signal = 11, + .periph_buses = PL08X_AHB2, + }, +}; + +struct pl08x_platform_data s3c64xx_dma1_plat_data = { + .memcpy_channel = { + .bus_id = "memcpy", + .cctl_memcpy = + (PL080_BSIZE_4 << PL080_CONTROL_SB_SIZE_SHIFT | + PL080_BSIZE_4 << PL080_CONTROL_DB_SIZE_SHIFT | + PL080_WIDTH_32BIT << PL080_CONTROL_SWIDTH_SHIFT | + PL080_WIDTH_32BIT << PL080_CONTROL_DWIDTH_SHIFT | + PL080_CONTROL_PROT_BUFF | PL080_CONTROL_PROT_CACHE | + PL080_CONTROL_PROT_SYS), + }, + .lli_buses = PL08X_AHB1, + .mem_buses = PL08X_AHB1, + .get_xfer_signal = pl08x_get_xfer_signal, + .put_xfer_signal = pl08x_put_xfer_signal, + .slave_channels = s3c64xx_dma1_info, + .num_slave_channels = ARRAY_SIZE(s3c64xx_dma1_info), +}; + +static AMBA_AHB_DEVICE(s3c64xx_dma1, "dma-pl080s.1", 0, + 0x75100000, {IRQ_DMA1}, &s3c64xx_dma1_plat_data); + +static int __init s3c64xx_pl080_init(void) +{ + /* Set all DMA configuration to be DMA, not SDMA */ + writel(0xffffff, S3C64XX_SDMA_SEL); + + if (of_have_populated_dt()) + return 0; + + amba_device_register(&s3c64xx_dma0_device, &iomem_resource); + amba_device_register(&s3c64xx_dma1_device, &iomem_resource); + + return 0; +} +arch_initcall(s3c64xx_pl080_init); diff --git a/arch/arm/plat-samsung/devs.c b/arch/arm/plat-samsung/devs.c index 99a3590..ac07e87 100644 --- a/arch/arm/plat-samsung/devs.c +++ b/arch/arm/plat-samsung/devs.c @@ -1468,6 +1468,8 @@ void __init s3c64xx_spi0_set_platdata(int (*cfg_gpio)(void), int src_clk_nr, pd.cfg_gpio = (cfg_gpio) ? cfg_gpio : s3c64xx_spi0_cfg_gpio; #if defined(CONFIG_PL330_DMA) pd.filter = pl330_filter; +#elif defined(CONFIG_S3C64XX_PL080) + pd.filter = pl08x_filter_id; #elif defined(CONFIG_S3C24XX_DMAC) pd.filter = s3c24xx_dma_filter; #endif @@ -1509,8 +1511,10 @@ void __init s3c64xx_spi1_set_platdata(int (*cfg_gpio)(void), int src_clk_nr, pd.num_cs = num_cs; pd.src_clk_nr = src_clk_nr; pd.cfg_gpio = (cfg_gpio) ? cfg_gpio : s3c64xx_spi1_cfg_gpio; -#ifdef CONFIG_PL330_DMA +#if defined(CONFIG_PL330_DMA) pd.filter = pl330_filter; +#elif defined(CONFIG_S3C64XX_PL080) + pd.filter = pl08x_filter_id; #endif s3c_set_platdata(&pd, sizeof(pd), &s3c64xx_device_spi1); @@ -1550,8 +1554,10 @@ void __init s3c64xx_spi2_set_platdata(int (*cfg_gpio)(void), int src_clk_nr, pd.num_cs = num_cs; pd.src_clk_nr = src_clk_nr; pd.cfg_gpio = (cfg_gpio) ? cfg_gpio : s3c64xx_spi2_cfg_gpio; -#ifdef CONFIG_PL330_DMA +#if defined(CONFIG_PL330_DMA) pd.filter = pl330_filter; +#elif defined(CONFIG_S3C64XX_PL080) + pd.filter = pl08x_filter_id; #endif s3c_set_platdata(&pd, sizeof(pd), &s3c64xx_device_spi2); diff --git a/arch/arm/plat-samsung/dma-ops.c b/arch/arm/plat-samsung/dma-ops.c index ec0d731..886326e 100644 --- a/arch/arm/plat-samsung/dma-ops.c +++ b/arch/arm/plat-samsung/dma-ops.c @@ -18,6 +18,12 @@ #include +#if defined(CONFIG_PL330_DMA) +#define dma_filter pl330_filter +#elif defined(CONFIG_S3C64XX_PL080) +#define dma_filter pl08x_filter_id +#endif + static unsigned samsung_dmadev_request(enum dma_ch dma_ch, struct samsung_dma_req *param, struct device *dev, char *ch_name) @@ -30,7 +36,7 @@ static unsigned samsung_dmadev_request(enum dma_ch dma_ch, if (dev->of_node) return (unsigned)dma_request_slave_channel(dev, ch_name); else - return (unsigned)dma_request_channel(mask, pl330_filter, + return (unsigned)dma_request_channel(mask, dma_filter, (void *)dma_ch); } -- cgit v0.10.2 From d37f7617bd677c46c49daa3c023920cb91fe14db Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 16 Oct 2013 21:10:55 +0200 Subject: clk: samsung: s3c64xx: Add aliases for DMA clocks This patch adds clkdev aliases for clocks used by PL08x DMA driver. Signed-off-by: Tomasz Figa Signed-off-by: Mark Brown diff --git a/drivers/clk/samsung/clk-s3c64xx.c b/drivers/clk/samsung/clk-s3c64xx.c index 7d2c842..06cf105 100644 --- a/drivers/clk/samsung/clk-s3c64xx.c +++ b/drivers/clk/samsung/clk-s3c64xx.c @@ -332,7 +332,9 @@ static struct samsung_clock_alias s3c64xx_clock_aliases[] = { ALIAS(HCLK_HSMMC0, "s3c-sdhci.0", "hsmmc"), ALIAS(HCLK_HSMMC0, "s3c-sdhci.0", "mmc_busclk.0"), ALIAS(HCLK_DMA1, NULL, "dma1"), + ALIAS(HCLK_DMA1, "dma-pl080s.1", "apb_pclk"), ALIAS(HCLK_DMA0, NULL, "dma0"), + ALIAS(HCLK_DMA0, "dma-pl080s.0", "apb_pclk"), ALIAS(HCLK_CAMIF, "s3c-camif", "camif"), ALIAS(HCLK_LCD, "s3c-fb", "lcd"), ALIAS(PCLK_SPI1, "s3c6410-spi.1", "spi"), -- cgit v0.10.2 From 15469ed37f8a9c004ac537495f9f7c51790a80c0 Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 16 Oct 2013 21:10:56 +0200 Subject: ARM: s3c64xx: Remove legacy DMA driver Since support for generic PL08x DMA engine driver has been added, there is no need to keep the old legacy driver, so this patch removes it. Signed-off-by: Tomasz Figa Signed-off-by: Mark Brown diff --git a/arch/arm/mach-s3c64xx/Kconfig b/arch/arm/mach-s3c64xx/Kconfig index d8e0288..7094bcc 100644 --- a/arch/arm/mach-s3c64xx/Kconfig +++ b/arch/arm/mach-s3c64xx/Kconfig @@ -22,11 +22,6 @@ config S3C64XX_PL080 select AMBA_PL08X select SAMSUNG_DMADEV -config S3C64XX_DMA - bool "S3C64XX DMA using legacy S3C DMA API" - select S3C_DMA - depends on !S3C64XX_PL080 - config S3C64XX_SETUP_SDHCI bool select S3C64XX_SETUP_SDHCI_GPIO diff --git a/arch/arm/mach-s3c64xx/Makefile b/arch/arm/mach-s3c64xx/Makefile index e8e9a46..58069a7 100644 --- a/arch/arm/mach-s3c64xx/Makefile +++ b/arch/arm/mach-s3c64xx/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_CPU_IDLE) += cpuidle.o # DMA support -obj-$(CONFIG_S3C64XX_DMA) += dma.o obj-$(CONFIG_S3C64XX_PL080) += pl080.o # Device support diff --git a/arch/arm/mach-s3c64xx/dma.c b/arch/arm/mach-s3c64xx/dma.c deleted file mode 100644 index 7e22c21..0000000 --- a/arch/arm/mach-s3c64xx/dma.c +++ /dev/null @@ -1,762 +0,0 @@ -/* linux/arch/arm/plat-s3c64xx/dma.c - * - * Copyright 2009 Openmoko, Inc. - * Copyright 2009 Simtec Electronics - * Ben Dooks - * http://armlinux.simtec.co.uk/ - * - * S3C64XX DMA core - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -/* - * NOTE: Code in this file is not used when booting with Device Tree support. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include - -#include "regs-sys.h" - -/* dma channel state information */ - -struct s3c64xx_dmac { - struct device dev; - struct clk *clk; - void __iomem *regs; - struct s3c2410_dma_chan *channels; - enum dma_ch chanbase; -}; - -/* pool to provide LLI buffers */ -static struct dma_pool *dma_pool; - -/* Debug configuration and code */ - -static unsigned char debug_show_buffs = 0; - -static void dbg_showchan(struct s3c2410_dma_chan *chan) -{ - pr_debug("DMA%d: %08x->%08x L %08x C %08x,%08x S %08x\n", - chan->number, - readl(chan->regs + PL080_CH_SRC_ADDR), - readl(chan->regs + PL080_CH_DST_ADDR), - readl(chan->regs + PL080_CH_LLI), - readl(chan->regs + PL080_CH_CONTROL), - readl(chan->regs + PL080S_CH_CONTROL2), - readl(chan->regs + PL080S_CH_CONFIG)); -} - -static void show_lli(struct pl080s_lli *lli) -{ - pr_debug("LLI[%p] %08x->%08x, NL %08x C %08x,%08x\n", - lli, lli->src_addr, lli->dst_addr, lli->next_lli, - lli->control0, lli->control1); -} - -static void dbg_showbuffs(struct s3c2410_dma_chan *chan) -{ - struct s3c64xx_dma_buff *ptr; - struct s3c64xx_dma_buff *end; - - pr_debug("DMA%d: buffs next %p, curr %p, end %p\n", - chan->number, chan->next, chan->curr, chan->end); - - ptr = chan->next; - end = chan->end; - - if (debug_show_buffs) { - for (; ptr != NULL; ptr = ptr->next) { - pr_debug("DMA%d: %08x ", - chan->number, ptr->lli_dma); - show_lli(ptr->lli); - } - } -} - -/* End of Debug */ - -static struct s3c2410_dma_chan *s3c64xx_dma_map_channel(unsigned int channel) -{ - struct s3c2410_dma_chan *chan; - unsigned int start, offs; - - start = 0; - - if (channel >= DMACH_PCM1_TX) - start = 8; - - for (offs = 0; offs < 8; offs++) { - chan = &s3c2410_chans[start + offs]; - if (!chan->in_use) - goto found; - } - - return NULL; - -found: - s3c_dma_chan_map[channel] = chan; - return chan; -} - -int s3c2410_dma_config(enum dma_ch channel, int xferunit) -{ - struct s3c2410_dma_chan *chan = s3c_dma_lookup_channel(channel); - - if (chan == NULL) - return -EINVAL; - - switch (xferunit) { - case 1: - chan->hw_width = 0; - break; - case 2: - chan->hw_width = 1; - break; - case 4: - chan->hw_width = 2; - break; - default: - printk(KERN_ERR "%s: illegal width %d\n", __func__, xferunit); - return -EINVAL; - } - - return 0; -} -EXPORT_SYMBOL(s3c2410_dma_config); - -static void s3c64xx_dma_fill_lli(struct s3c2410_dma_chan *chan, - struct pl080s_lli *lli, - dma_addr_t data, int size) -{ - dma_addr_t src, dst; - u32 control0, control1; - - switch (chan->source) { - case DMA_FROM_DEVICE: - src = chan->dev_addr; - dst = data; - control0 = PL080_CONTROL_SRC_AHB2; - control0 |= PL080_CONTROL_DST_INCR; - break; - - case DMA_TO_DEVICE: - src = data; - dst = chan->dev_addr; - control0 = PL080_CONTROL_DST_AHB2; - control0 |= PL080_CONTROL_SRC_INCR; - break; - default: - BUG(); - } - - /* note, we do not currently setup any of the burst controls */ - - control1 = size >> chan->hw_width; /* size in no of xfers */ - control0 |= PL080_CONTROL_PROT_SYS; /* always in priv. mode */ - control0 |= PL080_CONTROL_TC_IRQ_EN; /* always fire IRQ */ - control0 |= (u32)chan->hw_width << PL080_CONTROL_DWIDTH_SHIFT; - control0 |= (u32)chan->hw_width << PL080_CONTROL_SWIDTH_SHIFT; - - lli->src_addr = src; - lli->dst_addr = dst; - lli->next_lli = 0; - lli->control0 = control0; - lli->control1 = control1; -} - -static void s3c64xx_lli_to_regs(struct s3c2410_dma_chan *chan, - struct pl080s_lli *lli) -{ - void __iomem *regs = chan->regs; - - pr_debug("%s: LLI %p => regs\n", __func__, lli); - show_lli(lli); - - writel(lli->src_addr, regs + PL080_CH_SRC_ADDR); - writel(lli->dst_addr, regs + PL080_CH_DST_ADDR); - writel(lli->next_lli, regs + PL080_CH_LLI); - writel(lli->control0, regs + PL080_CH_CONTROL); - writel(lli->control1, regs + PL080S_CH_CONTROL2); -} - -static int s3c64xx_dma_start(struct s3c2410_dma_chan *chan) -{ - struct s3c64xx_dmac *dmac = chan->dmac; - u32 config; - u32 bit = chan->bit; - - dbg_showchan(chan); - - pr_debug("%s: clearing interrupts\n", __func__); - - /* clear interrupts */ - writel(bit, dmac->regs + PL080_TC_CLEAR); - writel(bit, dmac->regs + PL080_ERR_CLEAR); - - pr_debug("%s: starting channel\n", __func__); - - config = readl(chan->regs + PL080S_CH_CONFIG); - config |= PL080_CONFIG_ENABLE; - config &= ~PL080_CONFIG_HALT; - - pr_debug("%s: writing config %08x\n", __func__, config); - writel(config, chan->regs + PL080S_CH_CONFIG); - - return 0; -} - -static int s3c64xx_dma_stop(struct s3c2410_dma_chan *chan) -{ - u32 config; - int timeout; - - pr_debug("%s: stopping channel\n", __func__); - - dbg_showchan(chan); - - config = readl(chan->regs + PL080S_CH_CONFIG); - config |= PL080_CONFIG_HALT; - writel(config, chan->regs + PL080S_CH_CONFIG); - - timeout = 1000; - do { - config = readl(chan->regs + PL080S_CH_CONFIG); - pr_debug("%s: %d - config %08x\n", __func__, timeout, config); - if (config & PL080_CONFIG_ACTIVE) - udelay(10); - else - break; - } while (--timeout > 0); - - if (config & PL080_CONFIG_ACTIVE) { - printk(KERN_ERR "%s: channel still active\n", __func__); - return -EFAULT; - } - - config = readl(chan->regs + PL080S_CH_CONFIG); - config &= ~PL080_CONFIG_ENABLE; - writel(config, chan->regs + PL080S_CH_CONFIG); - - return 0; -} - -static inline void s3c64xx_dma_bufffdone(struct s3c2410_dma_chan *chan, - struct s3c64xx_dma_buff *buf, - enum s3c2410_dma_buffresult result) -{ - if (chan->callback_fn != NULL) - (chan->callback_fn)(chan, buf->pw, 0, result); -} - -static void s3c64xx_dma_freebuff(struct s3c64xx_dma_buff *buff) -{ - dma_pool_free(dma_pool, buff->lli, buff->lli_dma); - kfree(buff); -} - -static int s3c64xx_dma_flush(struct s3c2410_dma_chan *chan) -{ - struct s3c64xx_dma_buff *buff, *next; - u32 config; - - dbg_showchan(chan); - - pr_debug("%s: flushing channel\n", __func__); - - config = readl(chan->regs + PL080S_CH_CONFIG); - config &= ~PL080_CONFIG_ENABLE; - writel(config, chan->regs + PL080S_CH_CONFIG); - - /* dump all the buffers associated with this channel */ - - for (buff = chan->curr; buff != NULL; buff = next) { - next = buff->next; - pr_debug("%s: buff %p (next %p)\n", __func__, buff, buff->next); - - s3c64xx_dma_bufffdone(chan, buff, S3C2410_RES_ABORT); - s3c64xx_dma_freebuff(buff); - } - - chan->curr = chan->next = chan->end = NULL; - - return 0; -} - -int s3c2410_dma_ctrl(enum dma_ch channel, enum s3c2410_chan_op op) -{ - struct s3c2410_dma_chan *chan = s3c_dma_lookup_channel(channel); - - WARN_ON(!chan); - if (!chan) - return -EINVAL; - - switch (op) { - case S3C2410_DMAOP_START: - return s3c64xx_dma_start(chan); - - case S3C2410_DMAOP_STOP: - return s3c64xx_dma_stop(chan); - - case S3C2410_DMAOP_FLUSH: - return s3c64xx_dma_flush(chan); - - /* believe PAUSE/RESUME are no-ops */ - case S3C2410_DMAOP_PAUSE: - case S3C2410_DMAOP_RESUME: - case S3C2410_DMAOP_STARTED: - case S3C2410_DMAOP_TIMEOUT: - return 0; - } - - return -ENOENT; -} -EXPORT_SYMBOL(s3c2410_dma_ctrl); - -/* s3c2410_dma_enque - * - */ - -int s3c2410_dma_enqueue(enum dma_ch channel, void *id, - dma_addr_t data, int size) -{ - struct s3c2410_dma_chan *chan = s3c_dma_lookup_channel(channel); - struct s3c64xx_dma_buff *next; - struct s3c64xx_dma_buff *buff; - struct pl080s_lli *lli; - unsigned long flags; - int ret; - - WARN_ON(!chan); - if (!chan) - return -EINVAL; - - buff = kzalloc(sizeof(struct s3c64xx_dma_buff), GFP_ATOMIC); - if (!buff) { - printk(KERN_ERR "%s: no memory for buffer\n", __func__); - return -ENOMEM; - } - - lli = dma_pool_alloc(dma_pool, GFP_ATOMIC, &buff->lli_dma); - if (!lli) { - printk(KERN_ERR "%s: no memory for lli\n", __func__); - ret = -ENOMEM; - goto err_buff; - } - - pr_debug("%s: buff %p, dp %08x lli (%p, %08x) %d\n", - __func__, buff, data, lli, (u32)buff->lli_dma, size); - - buff->lli = lli; - buff->pw = id; - - s3c64xx_dma_fill_lli(chan, lli, data, size); - - local_irq_save(flags); - - if ((next = chan->next) != NULL) { - struct s3c64xx_dma_buff *end = chan->end; - struct pl080s_lli *endlli = end->lli; - - pr_debug("enquing onto channel\n"); - - end->next = buff; - endlli->next_lli = buff->lli_dma; - - if (chan->flags & S3C2410_DMAF_CIRCULAR) { - struct s3c64xx_dma_buff *curr = chan->curr; - lli->next_lli = curr->lli_dma; - } - - if (next == chan->curr) { - writel(buff->lli_dma, chan->regs + PL080_CH_LLI); - chan->next = buff; - } - - show_lli(endlli); - chan->end = buff; - } else { - pr_debug("enquing onto empty channel\n"); - - chan->curr = buff; - chan->next = buff; - chan->end = buff; - - s3c64xx_lli_to_regs(chan, lli); - } - - local_irq_restore(flags); - - show_lli(lli); - - dbg_showchan(chan); - dbg_showbuffs(chan); - return 0; - -err_buff: - kfree(buff); - return ret; -} - -EXPORT_SYMBOL(s3c2410_dma_enqueue); - - -int s3c2410_dma_devconfig(enum dma_ch channel, - enum dma_data_direction source, - unsigned long devaddr) -{ - struct s3c2410_dma_chan *chan = s3c_dma_lookup_channel(channel); - u32 peripheral; - u32 config = 0; - - pr_debug("%s: channel %d, source %d, dev %08lx, chan %p\n", - __func__, channel, source, devaddr, chan); - - WARN_ON(!chan); - if (!chan) - return -EINVAL; - - peripheral = (chan->peripheral & 0xf); - chan->source = source; - chan->dev_addr = devaddr; - - pr_debug("%s: peripheral %d\n", __func__, peripheral); - - switch (source) { - case DMA_FROM_DEVICE: - config = 2 << PL080_CONFIG_FLOW_CONTROL_SHIFT; - config |= peripheral << PL080_CONFIG_SRC_SEL_SHIFT; - break; - case DMA_TO_DEVICE: - config = 1 << PL080_CONFIG_FLOW_CONTROL_SHIFT; - config |= peripheral << PL080_CONFIG_DST_SEL_SHIFT; - break; - default: - printk(KERN_ERR "%s: bad source\n", __func__); - return -EINVAL; - } - - /* allow TC and ERR interrupts */ - config |= PL080_CONFIG_TC_IRQ_MASK; - config |= PL080_CONFIG_ERR_IRQ_MASK; - - pr_debug("%s: config %08x\n", __func__, config); - - writel(config, chan->regs + PL080S_CH_CONFIG); - - return 0; -} -EXPORT_SYMBOL(s3c2410_dma_devconfig); - - -int s3c2410_dma_getposition(enum dma_ch channel, - dma_addr_t *src, dma_addr_t *dst) -{ - struct s3c2410_dma_chan *chan = s3c_dma_lookup_channel(channel); - - WARN_ON(!chan); - if (!chan) - return -EINVAL; - - if (src != NULL) - *src = readl(chan->regs + PL080_CH_SRC_ADDR); - - if (dst != NULL) - *dst = readl(chan->regs + PL080_CH_DST_ADDR); - - return 0; -} -EXPORT_SYMBOL(s3c2410_dma_getposition); - -/* s3c2410_request_dma - * - * get control of an dma channel -*/ - -int s3c2410_dma_request(enum dma_ch channel, - struct s3c2410_dma_client *client, - void *dev) -{ - struct s3c2410_dma_chan *chan; - unsigned long flags; - - pr_debug("dma%d: s3c2410_request_dma: client=%s, dev=%p\n", - channel, client->name, dev); - - local_irq_save(flags); - - chan = s3c64xx_dma_map_channel(channel); - if (chan == NULL) { - local_irq_restore(flags); - return -EBUSY; - } - - dbg_showchan(chan); - - chan->client = client; - chan->in_use = 1; - chan->peripheral = channel; - chan->flags = 0; - - local_irq_restore(flags); - - /* need to setup */ - - pr_debug("%s: channel initialised, %p\n", __func__, chan); - - return chan->number | DMACH_LOW_LEVEL; -} - -EXPORT_SYMBOL(s3c2410_dma_request); - -/* s3c2410_dma_free - * - * release the given channel back to the system, will stop and flush - * any outstanding transfers, and ensure the channel is ready for the - * next claimant. - * - * Note, although a warning is currently printed if the freeing client - * info is not the same as the registrant's client info, the free is still - * allowed to go through. -*/ - -int s3c2410_dma_free(enum dma_ch channel, struct s3c2410_dma_client *client) -{ - struct s3c2410_dma_chan *chan = s3c_dma_lookup_channel(channel); - unsigned long flags; - - if (chan == NULL) - return -EINVAL; - - local_irq_save(flags); - - if (chan->client != client) { - printk(KERN_WARNING "dma%d: possible free from different client (channel %p, passed %p)\n", - channel, chan->client, client); - } - - /* sort out stopping and freeing the channel */ - - - chan->client = NULL; - chan->in_use = 0; - - if (!(channel & DMACH_LOW_LEVEL)) - s3c_dma_chan_map[channel] = NULL; - - local_irq_restore(flags); - - return 0; -} - -EXPORT_SYMBOL(s3c2410_dma_free); - -static irqreturn_t s3c64xx_dma_irq(int irq, void *pw) -{ - struct s3c64xx_dmac *dmac = pw; - struct s3c2410_dma_chan *chan; - enum s3c2410_dma_buffresult res; - u32 tcstat, errstat; - u32 bit; - int offs; - - tcstat = readl(dmac->regs + PL080_TC_STATUS); - errstat = readl(dmac->regs + PL080_ERR_STATUS); - - for (offs = 0, bit = 1; offs < 8; offs++, bit <<= 1) { - struct s3c64xx_dma_buff *buff; - - if (!(errstat & bit) && !(tcstat & bit)) - continue; - - chan = dmac->channels + offs; - res = S3C2410_RES_ERR; - - if (tcstat & bit) { - writel(bit, dmac->regs + PL080_TC_CLEAR); - res = S3C2410_RES_OK; - } - - if (errstat & bit) - writel(bit, dmac->regs + PL080_ERR_CLEAR); - - /* 'next' points to the buffer that is next to the - * currently active buffer. - * For CIRCULAR queues, 'next' will be same as 'curr' - * when 'end' is the active buffer. - */ - buff = chan->curr; - while (buff && buff != chan->next - && buff->next != chan->next) - buff = buff->next; - - if (!buff) - BUG(); - - if (buff == chan->next) - buff = chan->end; - - s3c64xx_dma_bufffdone(chan, buff, res); - - /* Free the node and update curr, if non-circular queue */ - if (!(chan->flags & S3C2410_DMAF_CIRCULAR)) { - chan->curr = buff->next; - s3c64xx_dma_freebuff(buff); - } - - /* Update 'next' */ - buff = chan->next; - if (chan->next == chan->end) { - chan->next = chan->curr; - if (!(chan->flags & S3C2410_DMAF_CIRCULAR)) - chan->end = NULL; - } else { - chan->next = buff->next; - } - } - - return IRQ_HANDLED; -} - -static struct bus_type dma_subsys = { - .name = "s3c64xx-dma", - .dev_name = "s3c64xx-dma", -}; - -static int s3c64xx_dma_init1(int chno, enum dma_ch chbase, - int irq, unsigned int base) -{ - struct s3c2410_dma_chan *chptr = &s3c2410_chans[chno]; - struct s3c64xx_dmac *dmac; - char clkname[16]; - void __iomem *regs; - void __iomem *regptr; - int err, ch; - - dmac = kzalloc(sizeof(struct s3c64xx_dmac), GFP_KERNEL); - if (!dmac) { - printk(KERN_ERR "%s: failed to alloc mem\n", __func__); - return -ENOMEM; - } - - dmac->dev.id = chno / 8; - dmac->dev.bus = &dma_subsys; - - err = device_register(&dmac->dev); - if (err) { - printk(KERN_ERR "%s: failed to register device\n", __func__); - goto err_alloc; - } - - regs = ioremap(base, 0x200); - if (!regs) { - printk(KERN_ERR "%s: failed to ioremap()\n", __func__); - err = -ENXIO; - goto err_dev; - } - - snprintf(clkname, sizeof(clkname), "dma%d", dmac->dev.id); - - dmac->clk = clk_get(NULL, clkname); - if (IS_ERR(dmac->clk)) { - printk(KERN_ERR "%s: failed to get clock %s\n", __func__, clkname); - err = PTR_ERR(dmac->clk); - goto err_map; - } - - clk_prepare_enable(dmac->clk); - - dmac->regs = regs; - dmac->chanbase = chbase; - dmac->channels = chptr; - - err = request_irq(irq, s3c64xx_dma_irq, 0, "DMA", dmac); - if (err < 0) { - printk(KERN_ERR "%s: failed to get irq\n", __func__); - goto err_clk; - } - - regptr = regs + PL080_Cx_BASE(0); - - for (ch = 0; ch < 8; ch++, chptr++) { - pr_debug("%s: registering DMA %d (%p)\n", - __func__, chno + ch, regptr); - - chptr->bit = 1 << ch; - chptr->number = chno + ch; - chptr->dmac = dmac; - chptr->regs = regptr; - regptr += PL080_Cx_STRIDE; - } - - /* for the moment, permanently enable the controller */ - writel(PL080_CONFIG_ENABLE, regs + PL080_CONFIG); - - printk(KERN_INFO "PL080: IRQ %d, at %p, channels %d..%d\n", - irq, regs, chno, chno+8); - - return 0; - -err_clk: - clk_disable_unprepare(dmac->clk); - clk_put(dmac->clk); -err_map: - iounmap(regs); -err_dev: - device_unregister(&dmac->dev); -err_alloc: - kfree(dmac); - return err; -} - -static int __init s3c64xx_dma_init(void) -{ - int ret; - - /* This driver is not supported when booting with device tree. */ - if (of_have_populated_dt()) - return -ENODEV; - - printk(KERN_INFO "%s: Registering DMA channels\n", __func__); - - dma_pool = dma_pool_create("DMA-LLI", NULL, sizeof(struct pl080s_lli), 16, 0); - if (!dma_pool) { - printk(KERN_ERR "%s: failed to create pool\n", __func__); - return -ENOMEM; - } - - ret = subsys_system_register(&dma_subsys, NULL); - if (ret) { - printk(KERN_ERR "%s: failed to create subsys\n", __func__); - return -ENOMEM; - } - - /* Set all DMA configuration to be DMA, not SDMA */ - writel(0xffffff, S3C64XX_SDMA_SEL); - - /* Register standard DMA controllers */ - s3c64xx_dma_init1(0, DMACH_UART0, IRQ_DMA0, 0x75000000); - s3c64xx_dma_init1(8, DMACH_PCM1_TX, IRQ_DMA1, 0x75100000); - - return 0; -} - -arch_initcall(s3c64xx_dma_init); diff --git a/arch/arm/mach-s3c64xx/include/mach/dma.h b/arch/arm/mach-s3c64xx/include/mach/dma.h index 26a6bc3..059b1fc 100644 --- a/arch/arm/mach-s3c64xx/include/mach/dma.h +++ b/arch/arm/mach-s3c64xx/include/mach/dma.h @@ -11,127 +11,6 @@ #ifndef __ASM_ARCH_DMA_H #define __ASM_ARCH_DMA_H __FILE__ -#ifdef CONFIG_S3C64XX_DMA - -#define S3C_DMA_CHANNELS (16) - -/* see mach-s3c2410/dma.h for notes on dma channel numbers */ - -/* Note, for the S3C64XX architecture we keep the DMACH_ - * defines in the order they are allocated to [S]DMA0/[S]DMA1 - * so that is easy to do DHACH_ -> DMA controller conversion - */ -enum dma_ch { - /* DMA0/SDMA0 */ - DMACH_UART0 = 0, - DMACH_UART0_SRC2, - DMACH_UART1, - DMACH_UART1_SRC2, - DMACH_UART2, - DMACH_UART2_SRC2, - DMACH_UART3, - DMACH_UART3_SRC2, - DMACH_PCM0_TX, - DMACH_PCM0_RX, - DMACH_I2S0_OUT, - DMACH_I2S0_IN, - DMACH_SPI0_TX, - DMACH_SPI0_RX, - DMACH_HSI_I2SV40_TX, - DMACH_HSI_I2SV40_RX, - - /* DMA1/SDMA1 */ - DMACH_PCM1_TX = 16, - DMACH_PCM1_RX, - DMACH_I2S1_OUT, - DMACH_I2S1_IN, - DMACH_SPI1_TX, - DMACH_SPI1_RX, - DMACH_AC97_PCMOUT, - DMACH_AC97_PCMIN, - DMACH_AC97_MICIN, - DMACH_PWM, - DMACH_IRDA, - DMACH_EXTERNAL, - DMACH_RES1, - DMACH_RES2, - DMACH_SECURITY_RX, /* SDMA1 only */ - DMACH_SECURITY_TX, /* SDMA1 only */ - DMACH_MAX /* the end */ -}; - -static inline bool samsung_dma_has_circular(void) -{ - return true; -} - -static inline bool samsung_dma_is_dmadev(void) -{ - return false; -} -#define S3C2410_DMAF_CIRCULAR (1 << 0) - -#include - -#define DMACH_LOW_LEVEL (1<<28) /* use this to specifiy hardware ch no */ - -struct s3c64xx_dma_buff; - -/** s3c64xx_dma_buff - S3C64XX DMA buffer descriptor - * @next: Pointer to next buffer in queue or ring. - * @pw: Client provided identifier - * @lli: Pointer to hardware descriptor this buffer is associated with. - * @lli_dma: Hardare address of the descriptor. - */ -struct s3c64xx_dma_buff { - struct s3c64xx_dma_buff *next; - - void *pw; - struct pl080s_lli *lli; - dma_addr_t lli_dma; -}; - -struct s3c64xx_dmac; - -struct s3c2410_dma_chan { - unsigned char number; /* number of this dma channel */ - unsigned char in_use; /* channel allocated */ - unsigned char bit; /* bit for enable/disable/etc */ - unsigned char hw_width; - unsigned char peripheral; - - unsigned int flags; - enum dma_data_direction source; - - - dma_addr_t dev_addr; - - struct s3c2410_dma_client *client; - struct s3c64xx_dmac *dmac; /* pointer to controller */ - - void __iomem *regs; - - /* cdriver callbacks */ - s3c2410_dma_cbfn_t callback_fn; /* buffer done callback */ - s3c2410_dma_opfn_t op_fn; /* channel op callback */ - - /* buffer list and information */ - struct s3c64xx_dma_buff *curr; /* current dma buffer */ - struct s3c64xx_dma_buff *next; /* next buffer to load */ - struct s3c64xx_dma_buff *end; /* end of queue */ - - /* note, when channel is running in circular mode, curr is the - * first buffer enqueued, end is the last and curr is where the - * last buffer-done event is set-at. The buffers are not freed - * and the last buffer hardware descriptor points back to the - * first. - */ -}; - -#include - -#else - #define S3C64XX_DMA_CHAN(name) ((unsigned long)(name)) /* DMA0/SDMA0 */ @@ -189,6 +68,4 @@ static inline bool samsung_dma_is_dmadev(void) #include #include -#endif - #endif /* __ASM_ARCH_IRQ_H */ -- cgit v0.10.2 From a7a996d19219424c8ca8f16474138b4ee170727f Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 16 Oct 2013 21:10:57 +0200 Subject: clk: samsung: s3c64xx: Remove clock aliases of old DMA driver Since the old DMA driver got removed, these aliases are no longer necessary. Signed-off-by: Tomasz Figa Signed-off-by: Mark Brown diff --git a/drivers/clk/samsung/clk-s3c64xx.c b/drivers/clk/samsung/clk-s3c64xx.c index 06cf105..8e27aee 100644 --- a/drivers/clk/samsung/clk-s3c64xx.c +++ b/drivers/clk/samsung/clk-s3c64xx.c @@ -331,9 +331,7 @@ static struct samsung_clock_alias s3c64xx_clock_aliases[] = { ALIAS(HCLK_HSMMC1, "s3c-sdhci.1", "mmc_busclk.0"), ALIAS(HCLK_HSMMC0, "s3c-sdhci.0", "hsmmc"), ALIAS(HCLK_HSMMC0, "s3c-sdhci.0", "mmc_busclk.0"), - ALIAS(HCLK_DMA1, NULL, "dma1"), ALIAS(HCLK_DMA1, "dma-pl080s.1", "apb_pclk"), - ALIAS(HCLK_DMA0, NULL, "dma0"), ALIAS(HCLK_DMA0, "dma-pl080s.0", "apb_pclk"), ALIAS(HCLK_CAMIF, "s3c-camif", "camif"), ALIAS(HCLK_LCD, "s3c-fb", "lcd"), -- cgit v0.10.2 From 3faecea70b0d6d050e0ae911032ec340341dc389 Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Wed, 16 Oct 2013 21:10:58 +0200 Subject: spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled The legacy S3C64xx DMA driver has been removed, DMA support on S3C64xx is provided only by the generic PL08x driver. This patch modifies the Kconfig entry of spi-s3c64xx driver, which relies on availability of DMA, to always select the S3C64XX_PL080 symbol. Signed-off-by: Tomasz Figa Signed-off-by: Mark Brown diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index 760d7b6..e2dd2fb 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -395,7 +395,7 @@ config SPI_S3C24XX_FIQ config SPI_S3C64XX tristate "Samsung S3C64XX series type SPI" depends on PLAT_SAMSUNG - select S3C64XX_DMA if ARCH_S3C64XX && !S3C64XX_PL080 + select S3C64XX_PL080 if ARCH_S3C64XX help SPI driver for Samsung S3C64XX and newer SoCs. -- cgit v0.10.2 From 93818c9a12dd38f2b32f960f979815ac2e15a176 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:42:51 +0000 Subject: ASoC: wm8990: Convet to module_i2c_driver() The device is I2C only (or at least current support is). Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 4f05fb8..33bec56 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1378,7 +1378,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .volatile_register = wm8990_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1420,29 +1419,8 @@ static struct i2c_driver wm8990_i2c_driver = { .remove = wm8990_i2c_remove, .id_table = wm8990_i2c_id, }; -#endif -static int __init wm8990_modinit(void) -{ - int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm8990_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8990 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm8990_modinit); - -static void __exit wm8990_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm8990_i2c_driver); -#endif -} -module_exit(wm8990_exit); +module_i2c_driver(wm8990_i2c_driver); MODULE_DESCRIPTION("ASoC WM8990 driver"); MODULE_AUTHOR("Liam Girdwood"); -- cgit v0.10.2 From f6b415b6065041c0970426cc8ac81a980b2998f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:44:56 +0000 Subject: ASoC: wm8990: Convert to table based control and DAPM init Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 33bec56..6ee1cf1 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -787,7 +787,7 @@ SND_SOC_DAPM_OUTPUT("RON"), SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8990_dapm_routes[] = { /* Make DACs turn on when playing even if not mixed into any outputs */ {"Internal DAC Sink", NULL, "Left DAC"}, {"Internal DAC Sink", NULL, "Right DAC"}, @@ -912,18 +912,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"RON", NULL, "RONMIX"}, }; -static int wm8990_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8990_dapm_widgets, - ARRAY_SIZE(wm8990_dapm_widgets)); - /* set up the WM8990 audio map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* PLL divisors */ struct _pll_div { u32 div2; @@ -1352,10 +1340,6 @@ static int wm8990_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_codec_controls(codec, wm8990_snd_controls, - ARRAY_SIZE(wm8990_snd_controls)); - wm8990_add_widgets(codec); - return 0; } @@ -1376,6 +1360,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8990_reg, .volatile_register = wm8990_volatile_register, + .controls = wm8990_snd_controls, + .num_controls = ARRAY_SIZE(wm8990_snd_controls), + .dapm_widgets = wm8990_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8990_dapm_widgets), + .dapm_routes = wm8990_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8990_dapm_routes), }; static int wm8990_i2c_probe(struct i2c_client *i2c, -- cgit v0.10.2 From d2fd5fe7ee3bc231e21aeb9ee120e0e61a79f8be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 14:25:04 +0000 Subject: ASoC: wm8990: Use supplies to manage input power Instead of using a fake register use a supply widget to manage the power for the inputs, this is more idiomatic and supports regmap conversion. Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 6ee1cf1..2261fe1 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -376,32 +376,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8990_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8990_INTDRIVBITS); - - if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) | - (1 << WM8990_AINLMUX_PWR_BIT))) { - reg |= WM8990_AINL_ENA; - } else { - reg &= ~WM8990_AINL_ENA; - } - - if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | - (1 << WM8990_AINRMUX_PWR_BIT))) { - reg |= WM8990_AINR_ENA; - } else { - reg &= ~WM8990_AINR_ENA; - } - snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -656,6 +630,11 @@ SND_SOC_DAPM_INPUT("RIN1"), SND_SOC_DAPM_INPUT("RIN2"), SND_SOC_DAPM_INPUT("Internal ADC Source"), +SND_SOC_DAPM_SUPPLY("INL", WM8990_POWER_MANAGEMENT_2, WM8990_AINL_ENA_BIT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8990_POWER_MANAGEMENT_2, WM8990_AINR_ENA_BIT, 0, + NULL, 0), + /* DACs */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2, WM8990_ADCL_ENA_BIT, 0), @@ -677,26 +656,20 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT, ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)), /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8990_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8990_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8990_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, - &wm8990_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0, &wm8990_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8990_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8990_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8990_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, - &wm8990_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0, &wm8990_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -796,6 +769,11 @@ static const struct snd_soc_dapm_route wm8990_dapm_routes[] = { {"Left ADC", NULL, "Internal ADC Source"}, {"Right ADC", NULL, "Internal ADC Source"}, + {"AINLMUX", NULL, "INL"}, + {"INMIXL", NULL, "INL"}, + {"AINRMUX", NULL, "INR"}, + {"INMIXR", NULL, "INR"}, + /* Input Side */ /* LIN12 PGA */ {"LIN12 PGA", "LIN1 Switch", "LIN1"}, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index 77c98a4..0e9c780 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -78,7 +78,6 @@ #define WM8990_PLL1 0x3C #define WM8990_PLL2 0x3D #define WM8990_PLL3 0x3E -#define WM8990_INTDRIVBITS 0x3F #define WM8990_EXT_ACCESS_ENA 0x75 #define WM8990_EXT_CTL1 0x7a @@ -818,14 +817,6 @@ */ #define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ -/* - * R63 (0x3F) - Internal Driver Bits - */ -#define WM8990_INMIXL_PWR_BIT 0 -#define WM8990_AINLMUX_PWR_BIT 1 -#define WM8990_INMIXR_PWR_BIT 2 -#define WM8990_AINRMUX_PWR_BIT 3 - #define WM8990_MCLK_DIV 0 #define WM8990_DACCLK_DIV 1 #define WM8990_ADCCLK_DIV 2 -- cgit v0.10.2 From 0112b62b12e18b883e1027689acab8eaa8830bac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 14:36:23 +0000 Subject: ASoC: wm8990: Convert to direct regmap API usage Signed-off-by: Mark Brown Acked-by: Charles Keepax diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 2261fe1..0ccd4d8 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -30,13 +31,12 @@ /* codec private data */ struct wm8990_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int sysclk; unsigned int pcmclk; }; -static int wm8990_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8990_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8990_RESET: @@ -46,71 +46,69 @@ static int wm8990_volatile_register(struct snd_soc_codec *codec, } } -static const u16 wm8990_reg[] = { - 0x8990, /* R0 - Reset */ - 0x0000, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x4050, /* R4 - Audio Interface (1) */ - 0x4000, /* R5 - Audio Interface (2) */ - 0x01C8, /* R6 - Clocking (1) */ - 0x0000, /* R7 - Clocking (2) */ - 0x0040, /* R8 - Audio Interface (3) */ - 0x0040, /* R9 - Audio Interface (4) */ - 0x0004, /* R10 - DAC CTRL */ - 0x00C0, /* R11 - Left DAC Digital Volume */ - 0x00C0, /* R12 - Right DAC Digital Volume */ - 0x0000, /* R13 - Digital Side Tone */ - 0x0100, /* R14 - ADC CTRL */ - 0x00C0, /* R15 - Left ADC Digital Volume */ - 0x00C0, /* R16 - Right ADC Digital Volume */ - 0x0000, /* R17 */ - 0x0000, /* R18 - GPIO CTRL 1 */ - 0x1000, /* R19 - GPIO1 & GPIO2 */ - 0x1010, /* R20 - GPIO3 & GPIO4 */ - 0x1010, /* R21 - GPIO5 & GPIO6 */ - 0x8000, /* R22 - GPIOCTRL 2 */ - 0x0800, /* R23 - GPIO_POL */ - 0x008B, /* R24 - Left Line Input 1&2 Volume */ - 0x008B, /* R25 - Left Line Input 3&4 Volume */ - 0x008B, /* R26 - Right Line Input 1&2 Volume */ - 0x008B, /* R27 - Right Line Input 3&4 Volume */ - 0x0000, /* R28 - Left Output Volume */ - 0x0000, /* R29 - Right Output Volume */ - 0x0066, /* R30 - Line Outputs Volume */ - 0x0022, /* R31 - Out3/4 Volume */ - 0x0079, /* R32 - Left OPGA Volume */ - 0x0079, /* R33 - Right OPGA Volume */ - 0x0003, /* R34 - Speaker Volume */ - 0x0003, /* R35 - ClassD1 */ - 0x0000, /* R36 */ - 0x0100, /* R37 - ClassD3 */ - 0x0079, /* R38 - ClassD4 */ - 0x0000, /* R39 - Input Mixer1 */ - 0x0000, /* R40 - Input Mixer2 */ - 0x0000, /* R41 - Input Mixer3 */ - 0x0000, /* R42 - Input Mixer4 */ - 0x0000, /* R43 - Input Mixer5 */ - 0x0000, /* R44 - Input Mixer6 */ - 0x0000, /* R45 - Output Mixer1 */ - 0x0000, /* R46 - Output Mixer2 */ - 0x0000, /* R47 - Output Mixer3 */ - 0x0000, /* R48 - Output Mixer4 */ - 0x0000, /* R49 - Output Mixer5 */ - 0x0000, /* R50 - Output Mixer6 */ - 0x0180, /* R51 - Out3/4 Mixer */ - 0x0000, /* R52 - Line Mixer1 */ - 0x0000, /* R53 - Line Mixer2 */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 - Additional Control */ - 0x0000, /* R56 - AntiPOP1 */ - 0x0000, /* R57 - AntiPOP2 */ - 0x0000, /* R58 - MICBIAS */ - 0x0000, /* R59 */ - 0x0008, /* R60 - PLL1 */ - 0x0031, /* R61 - PLL2 */ - 0x0026, /* R62 - PLL3 */ - 0x0000, /* R63 - Driver internal */ +static const struct reg_default wm8990_reg_defaults[] = { + { 1, 0x0000 }, /* R1 - Power Management (1) */ + { 2, 0x6000 }, /* R2 - Power Management (2) */ + { 3, 0x0000 }, /* R3 - Power Management (3) */ + { 4, 0x4050 }, /* R4 - Audio Interface (1) */ + { 5, 0x4000 }, /* R5 - Audio Interface (2) */ + { 6, 0x01C8 }, /* R6 - Clocking (1) */ + { 7, 0x0000 }, /* R7 - Clocking (2) */ + { 8, 0x0040 }, /* R8 - Audio Interface (3) */ + { 9, 0x0040 }, /* R9 - Audio Interface (4) */ + { 10, 0x0004 }, /* R10 - DAC CTRL */ + { 11, 0x00C0 }, /* R11 - Left DAC Digital Volume */ + { 12, 0x00C0 }, /* R12 - Right DAC Digital Volume */ + { 13, 0x0000 }, /* R13 - Digital Side Tone */ + { 14, 0x0100 }, /* R14 - ADC CTRL */ + { 15, 0x00C0 }, /* R15 - Left ADC Digital Volume */ + { 16, 0x00C0 }, /* R16 - Right ADC Digital Volume */ + + { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ + { 19, 0x1000 }, /* R19 - GPIO1 & GPIO2 */ + { 20, 0x1010 }, /* R20 - GPIO3 & GPIO4 */ + { 21, 0x1010 }, /* R21 - GPIO5 & GPIO6 */ + { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 23, 0x0800 }, /* R23 - GPIO_POL */ + { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ + { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ + { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ + { 27, 0x008B }, /* R27 - Right Line Input 3&4 Volume */ + { 28, 0x0000 }, /* R28 - Left Output Volume */ + { 29, 0x0000 }, /* R29 - Right Output Volume */ + { 30, 0x0066 }, /* R30 - Line Outputs Volume */ + { 31, 0x0022 }, /* R31 - Out3/4 Volume */ + { 32, 0x0079 }, /* R32 - Left OPGA Volume */ + { 33, 0x0079 }, /* R33 - Right OPGA Volume */ + { 34, 0x0003 }, /* R34 - Speaker Volume */ + { 35, 0x0003 }, /* R35 - ClassD1 */ + + { 37, 0x0100 }, /* R37 - ClassD3 */ + { 38, 0x0079 }, /* R38 - ClassD4 */ + { 39, 0x0000 }, /* R39 - Input Mixer1 */ + { 40, 0x0000 }, /* R40 - Input Mixer2 */ + { 41, 0x0000 }, /* R41 - Input Mixer3 */ + { 42, 0x0000 }, /* R42 - Input Mixer4 */ + { 43, 0x0000 }, /* R43 - Input Mixer5 */ + { 44, 0x0000 }, /* R44 - Input Mixer6 */ + { 45, 0x0000 }, /* R45 - Output Mixer1 */ + { 46, 0x0000 }, /* R46 - Output Mixer2 */ + { 47, 0x0000 }, /* R47 - Output Mixer3 */ + { 48, 0x0000 }, /* R48 - Output Mixer4 */ + { 49, 0x0000 }, /* R49 - Output Mixer5 */ + { 50, 0x0000 }, /* R50 - Output Mixer6 */ + { 51, 0x0180 }, /* R51 - Out3/4 Mixer */ + { 52, 0x0000 }, /* R52 - Line Mixer1 */ + { 53, 0x0000 }, /* R53 - Line Mixer2 */ + { 54, 0x0000 }, /* R54 - Speaker Mixer */ + { 55, 0x0000 }, /* R55 - Additional Control */ + { 56, 0x0000 }, /* R56 - AntiPOP1 */ + { 57, 0x0000 }, /* R57 - AntiPOP2 */ + { 58, 0x0000 }, /* R58 - MICBIAS */ + + { 60, 0x0008 }, /* R60 - PLL1 */ + { 61, 0x0031 }, /* R61 - PLL2 */ + { 62, 0x0026 }, /* R62 - PLL3 */ }; #define wm8990_reset(c) snd_soc_write(c, WM8990_RESET, 0) @@ -1114,6 +1112,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute) static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8990_priv *wm8990 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -1128,7 +1127,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8990->regmap); if (ret < 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); return ret; @@ -1226,7 +1225,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); - codec->cache_sync = 1; + regcache_mark_dirty(wm8990->regmap); break; } @@ -1295,7 +1294,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); return ret; @@ -1334,10 +1333,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .suspend = wm8990_suspend, .resume = wm8990_resume, .set_bias_level = wm8990_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8990_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8990_reg, - .volatile_register = wm8990_volatile_register, .controls = wm8990_snd_controls, .num_controls = ARRAY_SIZE(wm8990_snd_controls), .dapm_widgets = wm8990_dapm_widgets, @@ -1346,6 +1341,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .num_dapm_routes = ARRAY_SIZE(wm8990_dapm_routes), }; +static const struct regmap_config wm8990_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8990_PLL3, + .volatile_reg = wm8990_volatile_register, + .reg_defaults = wm8990_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8990_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- cgit v0.10.2 From 8778ac6be25abf0496fc614a3e77ad2ff8300353 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Nov 2013 15:55:06 +0100 Subject: ASoC: Fix build without CONFIG_GPIOLIB snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB explicitly with ifdef, and this causes build errors on some drivers such as: sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case instead. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f741cb..f7e1fac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -334,9 +334,7 @@ struct snd_soc_jack_pin; #include #include -#ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio; -#endif typedef int (*hw_write_t)(void *,const char* ,int); @@ -446,6 +444,17 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); +#else +static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + return 0; +} + +static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ +} #endif /* codec register bit access */ @@ -580,7 +589,6 @@ struct snd_soc_jack_zone { * to provide more complex checks (eg, reading an * ADC). */ -#ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio { unsigned int gpio; const char *name; @@ -594,7 +602,6 @@ struct snd_soc_jack_gpio { int (*jack_status_check)(void); }; -#endif struct snd_soc_jack { struct mutex mutex; -- cgit v0.10.2 From d44008b358588cf6fcc74716b50584a8e59cbe65 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:48:31 -0200 Subject: ASoC: wm8995: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index da2899e6..4300caf 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2293,7 +2293,7 @@ static struct spi_driver wm8995_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8995_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2350,7 +2350,7 @@ static int __init wm8995_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8995_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8995 I2C driver: %d\n", @@ -2371,7 +2371,7 @@ module_init(wm8995_modinit); static void __exit wm8995_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8995_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From c6aeb7de226dd08ad9b343fc6cbaf2ff36f69c84 Mon Sep 17 00:00:00 2001 From: Florian Meier Date: Fri, 22 Nov 2013 16:24:08 +0100 Subject: ASoC: Add support for BCM2835 This driver adds support for digital audio (I2S) for the BCM2835 SoC that is used by the Raspberry Pi. External audio codecs can be connected to the Raspberry Pi via P5 header. It relies on cyclic DMA engine support for BCM2835. Signed-off-by: Florian Meier Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt new file mode 100644 index 0000000..65783de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt @@ -0,0 +1,25 @@ +* Broadcom BCM2835 SoC I2S/PCM module + +Required properties: +- compatible: "brcm,bcm2835-i2s" +- reg: A list of base address and size entries: + * The first entry should cover the PCM registers + * The second entry should cover the PCM clock registers +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = <0x7e203000 0x20>, + <0x7e101098 0x02>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b84..a5e3a70 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/bcm/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e701..b52d4aa 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += bcm/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig new file mode 100644 index 0000000..3d82a29 --- /dev/null +++ b/sound/soc/bcm/Kconfig @@ -0,0 +1,10 @@ +config SND_BCM2835_SOC_I2S + tristate "SoC Audio support for the Broadcom BCM2835 I2S module" + depends on ARCH_BCM2835 || COMPILE_TEST + select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to + the BCM2835 I2S interface. You will also need + to select the audio interfaces to support below. diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile new file mode 100644 index 0000000..bc816b7 --- /dev/null +++ b/sound/soc/bcm/Makefile @@ -0,0 +1,5 @@ +# BCM2835 Platform Support +snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o + +obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o + diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c new file mode 100644 index 0000000..f49b007 --- /dev/null +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -0,0 +1,886 @@ +/* + * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC + * + * Author: Florian Meier + * Copyright 2013 + * + * Based on + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * Vladimir Barinov, + * Copyright (C) 2007 MontaVista Software, Inc., + * + * OMAP ALSA SoC DAI driver using McBSP port + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula + * Peter Ujfalusi + * + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * Author: Timur Tabi + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +/* Clock registers */ +#define BCM2835_CLK_PCMCTL_REG 0x00 +#define BCM2835_CLK_PCMDIV_REG 0x04 + +/* Clock register settings */ +#define BCM2835_CLK_PASSWD (0x5a000000) +#define BCM2835_CLK_PASSWD_MASK (0xff000000) +#define BCM2835_CLK_MASH(v) ((v) << 9) +#define BCM2835_CLK_FLIP BIT(8) +#define BCM2835_CLK_BUSY BIT(7) +#define BCM2835_CLK_KILL BIT(5) +#define BCM2835_CLK_ENAB BIT(4) +#define BCM2835_CLK_SRC(v) (v) + +#define BCM2835_CLK_SHIFT (12) +#define BCM2835_CLK_DIVI(v) ((v) << BCM2835_CLK_SHIFT) +#define BCM2835_CLK_DIVF(v) (v) +#define BCM2835_CLK_DIVF_MASK (0xFFF) + +enum { + BCM2835_CLK_MASH_0 = 0, + BCM2835_CLK_MASH_1, + BCM2835_CLK_MASH_2, + BCM2835_CLK_MASH_3, +}; + +enum { + BCM2835_CLK_SRC_GND = 0, + BCM2835_CLK_SRC_OSC, + BCM2835_CLK_SRC_DBG0, + BCM2835_CLK_SRC_DBG1, + BCM2835_CLK_SRC_PLLA, + BCM2835_CLK_SRC_PLLC, + BCM2835_CLK_SRC_PLLD, + BCM2835_CLK_SRC_HDMI, +}; + +/* Most clocks are not useable (freq = 0) */ +static const unsigned int bcm2835_clk_freq[BCM2835_CLK_SRC_HDMI+1] = { + [BCM2835_CLK_SRC_GND] = 0, + [BCM2835_CLK_SRC_OSC] = 19200000, + [BCM2835_CLK_SRC_DBG0] = 0, + [BCM2835_CLK_SRC_DBG1] = 0, + [BCM2835_CLK_SRC_PLLA] = 0, + [BCM2835_CLK_SRC_PLLC] = 0, + [BCM2835_CLK_SRC_PLLD] = 500000000, + [BCM2835_CLK_SRC_HDMI] = 0, +}; + +/* I2S registers */ +#define BCM2835_I2S_CS_A_REG 0x00 +#define BCM2835_I2S_FIFO_A_REG 0x04 +#define BCM2835_I2S_MODE_A_REG 0x08 +#define BCM2835_I2S_RXC_A_REG 0x0c +#define BCM2835_I2S_TXC_A_REG 0x10 +#define BCM2835_I2S_DREQ_A_REG 0x14 +#define BCM2835_I2S_INTEN_A_REG 0x18 +#define BCM2835_I2S_INTSTC_A_REG 0x1c +#define BCM2835_I2S_GRAY_REG 0x20 + +/* I2S register settings */ +#define BCM2835_I2S_STBY BIT(25) +#define BCM2835_I2S_SYNC BIT(24) +#define BCM2835_I2S_RXSEX BIT(23) +#define BCM2835_I2S_RXF BIT(22) +#define BCM2835_I2S_TXE BIT(21) +#define BCM2835_I2S_RXD BIT(20) +#define BCM2835_I2S_TXD BIT(19) +#define BCM2835_I2S_RXR BIT(18) +#define BCM2835_I2S_TXW BIT(17) +#define BCM2835_I2S_CS_RXERR BIT(16) +#define BCM2835_I2S_CS_TXERR BIT(15) +#define BCM2835_I2S_RXSYNC BIT(14) +#define BCM2835_I2S_TXSYNC BIT(13) +#define BCM2835_I2S_DMAEN BIT(9) +#define BCM2835_I2S_RXTHR(v) ((v) << 7) +#define BCM2835_I2S_TXTHR(v) ((v) << 5) +#define BCM2835_I2S_RXCLR BIT(4) +#define BCM2835_I2S_TXCLR BIT(3) +#define BCM2835_I2S_TXON BIT(2) +#define BCM2835_I2S_RXON BIT(1) +#define BCM2835_I2S_EN (1) + +#define BCM2835_I2S_CLKDIS BIT(28) +#define BCM2835_I2S_PDMN BIT(27) +#define BCM2835_I2S_PDME BIT(26) +#define BCM2835_I2S_FRXP BIT(25) +#define BCM2835_I2S_FTXP BIT(24) +#define BCM2835_I2S_CLKM BIT(23) +#define BCM2835_I2S_CLKI BIT(22) +#define BCM2835_I2S_FSM BIT(21) +#define BCM2835_I2S_FSI BIT(20) +#define BCM2835_I2S_FLEN(v) ((v) << 10) +#define BCM2835_I2S_FSLEN(v) (v) + +#define BCM2835_I2S_CHWEX BIT(15) +#define BCM2835_I2S_CHEN BIT(14) +#define BCM2835_I2S_CHPOS(v) ((v) << 4) +#define BCM2835_I2S_CHWID(v) (v) +#define BCM2835_I2S_CH1(v) ((v) << 16) +#define BCM2835_I2S_CH2(v) (v) + +#define BCM2835_I2S_TX_PANIC(v) ((v) << 24) +#define BCM2835_I2S_RX_PANIC(v) ((v) << 16) +#define BCM2835_I2S_TX(v) ((v) << 8) +#define BCM2835_I2S_RX(v) (v) + +#define BCM2835_I2S_INT_RXERR BIT(3) +#define BCM2835_I2S_INT_TXERR BIT(2) +#define BCM2835_I2S_INT_RXR BIT(1) +#define BCM2835_I2S_INT_TXW BIT(0) + +/* I2S DMA interface */ +/* FIXME: Needs IOMMU support */ +#define BCM2835_VCMMU_SHIFT (0x7E000000 - 0x20000000) + +/* General device struct */ +struct bcm2835_i2s_dev { + struct device *dev; + struct snd_dmaengine_dai_dma_data dma_data[2]; + unsigned int fmt; + unsigned int bclk_ratio; + + struct regmap *i2s_regmap; + struct regmap *clk_regmap; +}; + +static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) +{ + /* Start the clock if in master mode */ + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + break; + default: + break; + } +} + +static void bcm2835_i2s_stop_clock(struct bcm2835_i2s_dev *dev) +{ + uint32_t clkreg; + int timeout = 1000; + + /* Stop clock */ + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD); + + /* Wait for the BUSY flag going down */ + while (--timeout) { + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + if (!(clkreg & BCM2835_CLK_BUSY)) + break; + } + + if (!timeout) { + /* KILL the clock */ + dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n"); + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD_MASK, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD); + } +} + +static void bcm2835_i2s_clear_fifos(struct bcm2835_i2s_dev *dev, + bool tx, bool rx) +{ + int timeout = 1000; + uint32_t syncval; + uint32_t csreg; + uint32_t i2s_active_state; + uint32_t clkreg; + uint32_t clk_active_state; + uint32_t off; + uint32_t clr; + + off = tx ? BCM2835_I2S_TXON : 0; + off |= rx ? BCM2835_I2S_RXON : 0; + + clr = tx ? BCM2835_I2S_TXCLR : 0; + clr |= rx ? BCM2835_I2S_RXCLR : 0; + + /* Backup the current state */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + i2s_active_state = csreg & (BCM2835_I2S_RXON | BCM2835_I2S_TXON); + + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + clk_active_state = clkreg & BCM2835_CLK_ENAB; + + /* Start clock if not running */ + if (!clk_active_state) { + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + } + + /* Stop I2S module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, off, 0); + + /* + * Clear the FIFOs + * Requires at least 2 PCM clock cycles to take effect + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, clr, clr); + + /* Wait for 2 PCM clock cycles */ + + /* + * Toggle the SYNC flag. After 2 PCM clock cycles it can be read back + * FIXME: This does not seem to work for slave mode! + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &syncval); + syncval &= BCM2835_I2S_SYNC; + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_SYNC, ~syncval); + + /* Wait for the SYNC flag changing it's state */ + while (--timeout) { + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + if ((csreg & BCM2835_I2S_SYNC) != syncval) + break; + } + + if (!timeout) + dev_err(dev->dev, "I2S SYNC error!\n"); + + /* Stop clock if it was not running before */ + if (!clk_active_state) + bcm2835_i2s_stop_clock(dev); + + /* Restore I2S state */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXON | BCM2835_I2S_TXON, i2s_active_state); +} + +static int bcm2835_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->fmt = fmt; + return 0; +} + +static int bcm2835_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->bclk_ratio = ratio; + return 0; +} + +static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + unsigned int sampling_rate = params_rate(params); + unsigned int data_length, data_delay, bclk_ratio; + unsigned int ch1pos, ch2pos, mode, format; + unsigned int mash = BCM2835_CLK_MASH_1; + unsigned int divi, divf, target_frequency; + int clk_src = -1; + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + bool bit_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM); + + bool frame_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBM_CFS); + uint32_t csreg; + + /* + * If a stream is already enabled, + * the registers are already set properly. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + + if (csreg & (BCM2835_I2S_TXON | BCM2835_I2S_RXON)) + return 0; + + /* + * Adjust the data length according to the format. + * We prefill the half frame length with an integer + * divider of 2400 as explained at the clock settings. + * Maybe it is overwritten there, if the Integer mode + * does not apply. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_length = 16; + bclk_ratio = 40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + data_length = 32; + bclk_ratio = 80; + break; + default: + return -EINVAL; + } + + /* If bclk_ratio already set, use that one. */ + if (dev->bclk_ratio) + bclk_ratio = dev->bclk_ratio; + + /* + * Clock Settings + * + * The target frequency of the bit clock is + * sampling rate * frame length + * + * Integer mode: + * Sampling rates that are multiples of 8000 kHz + * can be driven by the oscillator of 19.2 MHz + * with an integer divider as long as the frame length + * is an integer divider of 19200000/8000=2400 as set up above. + * This is no longer possible if the sampling rate + * is too high (e.g. 192 kHz), because the oscillator is too slow. + * + * MASH mode: + * For all other sampling rates, it is not possible to + * have an integer divider. Approximate the clock + * with the MASH module that induces a slight frequency + * variance. To minimize that it is best to have the fastest + * clock here. That is PLLD with 500 MHz. + */ + target_frequency = sampling_rate * bclk_ratio; + clk_src = BCM2835_CLK_SRC_OSC; + mash = BCM2835_CLK_MASH_0; + + if (bcm2835_clk_freq[clk_src] % target_frequency == 0 + && bit_master && frame_master) { + divi = bcm2835_clk_freq[clk_src] / target_frequency; + divf = 0; + } else { + uint64_t dividend; + + if (!dev->bclk_ratio) { + /* + * Overwrite bclk_ratio, because the + * above trick is not needed or can + * not be used. + */ + bclk_ratio = 2 * data_length; + } + + target_frequency = sampling_rate * bclk_ratio; + + clk_src = BCM2835_CLK_SRC_PLLD; + mash = BCM2835_CLK_MASH_1; + + dividend = bcm2835_clk_freq[clk_src]; + dividend <<= BCM2835_CLK_SHIFT; + do_div(dividend, target_frequency); + divi = dividend >> BCM2835_CLK_SHIFT; + divf = dividend & BCM2835_CLK_DIVF_MASK; + } + + /* Set clock divider */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMDIV_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_DIVI(divi) + | BCM2835_CLK_DIVF(divf)); + + /* Setup clock, but don't start it yet */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_MASH(mash) + | BCM2835_CLK_SRC(clk_src)); + + /* Setup the frame format */ + format = BCM2835_I2S_CHEN; + + if (data_length > 24) + format |= BCM2835_I2S_CHWEX; + + format |= BCM2835_I2S_CHWID((data_length-8)&0xf); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data_delay = 1; + break; + default: + /* + * TODO + * Others are possible but are not implemented at the moment. + */ + dev_err(dev->dev, "%s:bad format\n", __func__); + return -EINVAL; + } + + ch1pos = data_delay; + ch2pos = bclk_ratio / 2 + data_delay; + + switch (params_channels(params)) { + case 2: + format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); + format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); + format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); + break; + default: + return -EINVAL; + } + + /* + * Set format for both streams. + * We cannot set another frame length + * (and therefore word length) anyway, + * so the format will be the same. + */ + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, format); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, format); + + /* Setup the I2S mode */ + mode = 0; + + if (data_length <= 16) { + /* + * Use frame packed mode (2 channels per 32 bit word) + * We cannot set another frame length in the second stream + * (and therefore word length) anyway, + * so the format will be the same. + */ + mode |= BCM2835_I2S_FTXP | BCM2835_I2S_FRXP; + } + + mode |= BCM2835_I2S_FLEN(bclk_ratio - 1); + mode |= BCM2835_I2S_FSLEN(bclk_ratio / 2); + + /* Master or slave? */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* CPU is master */ + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* + * CODEC is bit clock master + * CPU is frame master + */ + mode |= BCM2835_I2S_CLKM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* + * CODEC is frame master + * CPU is bit clock master + */ + mode |= BCM2835_I2S_FSM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CODEC is master */ + mode |= BCM2835_I2S_CLKM; + mode |= BCM2835_I2S_FSM; + break; + default: + dev_err(dev->dev, "%s:bad master\n", __func__); + return -EINVAL; + } + + /* + * Invert clocks? + * + * The BCM approach seems to be inverted to the classical I2S approach. + */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* None. Therefore, both for BCM */ + mode |= BCM2835_I2S_CLKI; + mode |= BCM2835_I2S_FSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Both. Therefore, none for BCM */ + break; + case SND_SOC_DAIFMT_NB_IF: + /* + * Invert only frame sync. Therefore, + * invert only bit clock for BCM + */ + mode |= BCM2835_I2S_CLKI; + break; + case SND_SOC_DAIFMT_IB_NF: + /* + * Invert only bit clock. Therefore, + * invert only frame sync for BCM + */ + mode |= BCM2835_I2S_FSI; + break; + default: + return -EINVAL; + } + + regmap_write(dev->i2s_regmap, BCM2835_I2S_MODE_A_REG, mode); + + /* Setup the DMA parameters */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXTHR(1) + | BCM2835_I2S_TXTHR(1) + | BCM2835_I2S_DMAEN, 0xffffffff); + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_DREQ_A_REG, + BCM2835_I2S_TX_PANIC(0x10) + | BCM2835_I2S_RX_PANIC(0x30) + | BCM2835_I2S_TX(0x30) + | BCM2835_I2S_RX(0x20), 0xffffffff); + + /* Clear FIFOs */ + bcm2835_i2s_clear_fifos(dev, true, true); + + return 0; +} + +static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t cs_reg; + + bcm2835_i2s_start_clock(dev); + + /* + * Clear both FIFOs if the one that should be started + * is not empty at the moment. This should only happen + * after overrun. Otherwise, hw_params would have cleared + * the FIFO. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &cs_reg); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + && !(cs_reg & BCM2835_I2S_TXE)) + bcm2835_i2s_clear_fifos(dev, true, false); + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + && (cs_reg & BCM2835_I2S_RXD)) + bcm2835_i2s_clear_fifos(dev, false, true); + + return 0; +} + +static void bcm2835_i2s_stop(struct bcm2835_i2s_dev *dev, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + uint32_t mask; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, 0); + + /* Stop also the clock when not SND_SOC_DAIFMT_CONT */ + if (!dai->active && !(dev->fmt & SND_SOC_DAIFMT_CONT)) + bcm2835_i2s_stop_clock(dev); +} + +static int bcm2835_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t mask; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + bcm2835_i2s_start_clock(dev); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, mask); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + bcm2835_i2s_stop(dev, substream, dai); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int bcm2835_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (dai->active) + return 0; + + /* Should this still be running stop it */ + bcm2835_i2s_stop_clock(dev); + + /* Enable PCM block */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, BCM2835_I2S_EN); + + /* + * Disable STBY. + * Requires at least 4 PCM clock cycles to take effect. + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_STBY, BCM2835_I2S_STBY); + + return 0; +} + +static void bcm2835_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + bcm2835_i2s_stop(dev, substream, dai); + + /* If both streams are stopped, disable module and clock */ + if (dai->active) + return; + + /* Disable the module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, 0); + + /* + * Stopping clock is necessary, because stop does + * not stop the clock when SND_SOC_DAIFMT_CONT + */ + bcm2835_i2s_stop_clock(dev); +} + +static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = { + .startup = bcm2835_i2s_startup, + .shutdown = bcm2835_i2s_shutdown, + .prepare = bcm2835_i2s_prepare, + .trigger = bcm2835_i2s_trigger, + .hw_params = bcm2835_i2s_hw_params, + .set_fmt = bcm2835_i2s_set_dai_fmt, + .set_bclk_ratio = bcm2835_i2s_set_dai_bclk_ratio +}; + +static int bcm2835_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, + &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK], + &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]); + + return 0; +} + +static struct snd_soc_dai_driver bcm2835_i2s_dai = { + .name = "bcm2835-i2s", + .probe = bcm2835_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .ops = &bcm2835_i2s_dai_ops, + .symmetric_rates = 1 +}; + +static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_CS_A_REG: + case BCM2835_I2S_FIFO_A_REG: + case BCM2835_I2S_INTSTC_A_REG: + case BCM2835_I2S_GRAY_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_FIFO_A_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_clk_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_CLK_PCMCTL_REG: + return true; + default: + return false; + }; +} + +static const struct regmap_config bcm2835_regmap_config[] = { + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_I2S_GRAY_REG, + .precious_reg = bcm2835_i2s_precious_reg, + .volatile_reg = bcm2835_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_CLK_PCMDIV_REG, + .volatile_reg = bcm2835_clk_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, +}; + +static const struct snd_soc_component_driver bcm2835_i2s_component = { + .name = "bcm2835-i2s-comp", +}; + +static int bcm2835_i2s_probe(struct platform_device *pdev) +{ + struct bcm2835_i2s_dev *dev; + int i; + int ret; + struct regmap *regmap[2]; + struct resource *mem[2]; + + /* Request both ioareas */ + for (i = 0; i <= 1; i++) { + void __iomem *base; + + mem[i] = platform_get_resource(pdev, IORESOURCE_MEM, i); + base = devm_ioremap_resource(&pdev->dev, mem[i]); + if (IS_ERR(base)) + return PTR_ERR(base); + + regmap[i] = devm_regmap_init_mmio(&pdev->dev, base, + &bcm2835_regmap_config[i]); + if (IS_ERR(regmap[i])) + return PTR_ERR(regmap[i]); + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dev->i2s_regmap = regmap[0]; + dev->clk_regmap = regmap[1]; + + /* Set the DMA address */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + /* Set the bus width */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + + /* Set burst */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + + /* BCLK ratio - use default */ + dev->bclk_ratio = 0; + + /* Store the pdev */ + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &bcm2835_i2s_component, &bcm2835_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + return ret; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct of_device_id bcm2835_i2s_of_match[] = { + { .compatible = "brcm,bcm2835-i2s", }, + {}, +}; + +static int bcm2835_i2s_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2835_i2s_driver = { + .probe = bcm2835_i2s_probe, + .remove = bcm2835_i2s_remove, + .driver = { + .name = "bcm2835-i2s", + .owner = THIS_MODULE, + .of_match_table = bcm2835_i2s_of_match, + }, +}; + +module_platform_driver(bcm2835_i2s_driver); + +MODULE_ALIAS("platform:bcm2835-i2s"); +MODULE_DESCRIPTION("BCM2835 I2S interface"); +MODULE_AUTHOR("Florian Meier "); +MODULE_LICENSE("GPL v2"); -- cgit v0.10.2 From 516ea4b584332f511d3bf1b98ceabd974b1a2313 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:38 -0200 Subject: ASoC: cs4271: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f6e9534..ce05fd9 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -675,7 +675,7 @@ static struct spi_driver cs4271_spi_driver = { }; #endif /* defined(CONFIG_SPI_MASTER) */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static const struct i2c_device_id cs4271_i2c_id[] = { {"cs4271", 0}, {} @@ -728,7 +728,7 @@ static struct i2c_driver cs4271_i2c_driver = { .probe = cs4271_i2c_probe, .remove = cs4271_i2c_remove, }; -#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ +#endif /* IS_ENABLED(CONFIG_I2C) */ /* * We only register our serial bus driver here without @@ -741,7 +741,7 @@ static int __init cs4271_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&cs4271_i2c_driver); if (ret) { pr_err("Failed to register CS4271 I2C driver: %d\n", ret); @@ -767,7 +767,7 @@ static void __exit cs4271_modexit(void) spi_unregister_driver(&cs4271_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&cs4271_i2c_driver); #endif } -- cgit v0.10.2 From 25c1a63f43ca40f1581c076b7f7618297ef1cbba Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:40 -0200 Subject: ASoC: da7210: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 9c12314..8166dcb 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1188,7 +1188,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static struct reg_default da7210_regmap_i2c_patch[] = { @@ -1362,7 +1362,7 @@ static struct spi_driver da7210_spi_driver = { static int __init da7210_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&da7210_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) @@ -1378,7 +1378,7 @@ module_init(da7210_modinit); static void __exit da7210_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&da7210_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From e42644c64c0de145ab2041ceb904e44421dd6794 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:42 -0200 Subject: ASoC: ssm2602: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 492644e..480074d 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -730,7 +730,7 @@ static struct spi_driver ssm2602_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) /* * ssm2602 2 wire address is determined by GPIO5 * state during powerup. @@ -797,7 +797,7 @@ static int __init ssm2602_modinit(void) return ret; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&ssm2602_i2c_driver); if (ret) return ret; @@ -813,7 +813,7 @@ static void __exit ssm2602_exit(void) spi_unregister_driver(&ssm2602_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&ssm2602_i2c_driver); #endif } -- cgit v0.10.2 From b65ab73e5d624eb4a88bc6094a3627007cb92500 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:43 -0200 Subject: ASoC: wm8731: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c..6117107 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -732,7 +732,7 @@ static struct spi_driver wm8731_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8731_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -791,7 +791,7 @@ static struct i2c_driver wm8731_i2c_driver = { static int __init wm8731_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8731_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n", @@ -811,7 +811,7 @@ module_init(wm8731_modinit); static void __exit wm8731_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8731_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 26090a834b49673945458b185be0afa03c2737fe Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:45 -0200 Subject: ASoC: wm8741: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b18813c..2895c8d 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -500,7 +500,7 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -617,7 +617,7 @@ static int __init wm8741_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8741_i2c_driver); if (ret != 0) pr_err("Failed to register WM8741 I2C driver: %d\n", ret); @@ -639,7 +639,7 @@ static void __exit wm8741_exit(void) #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8741_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8741_i2c_driver); #endif } -- cgit v0.10.2 From 9ea6fbc66d15c83089e177b445872a9ba40f125d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:46 -0200 Subject: ASoC: wm8750: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 50d5ff6..78616a6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -816,7 +816,7 @@ static struct spi_driver wm8750_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -868,7 +868,7 @@ static struct i2c_driver wm8750_i2c_driver = { static int __init wm8750_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8750_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8750 I2C driver: %d\n", @@ -888,7 +888,7 @@ module_init(wm8750_modinit); static void __exit wm8750_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 2c4864334c4d9a23fa810638ad27e80ea0ceb9a4 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:47 -0200 Subject: ASoC: wm8753: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d96ebf5..be85da9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1596,7 +1596,7 @@ static struct spi_driver wm8753_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8753_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1653,7 +1653,7 @@ static struct i2c_driver wm8753_i2c_driver = { static int __init wm8753_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8753_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8753 I2C driver: %d\n", @@ -1673,7 +1673,7 @@ module_init(wm8753_modinit); static void __exit wm8753_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8753_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From b888edbc68fbace3101cb092c6910476e85ae922 Mon Sep 17 00:00:00 2001 From: wangbiao Date: Fri, 22 Nov 2013 10:44:30 +0800 Subject: ASoC: wm8994: Move DCS done IRQ request later once code return from request_threaded_irq, irq was setup enabled by default, but completion var dcs_done not got initialized yet, if then a dcs done irq was raised, system will got hung as the sync mechanism is invalid now. so this patch move dcs done irq request to the end of initialization of completion. Signed-off-by: wang, biao Signed-off-by: Zhang, Di Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 86426a1..b9be9cb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4077,12 +4077,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); - ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, - wm_hubs_dcs_done, "DC servo done", - &wm8994->hubs); - if (ret == 0) - wm8994->hubs.dcs_done_irq = true; - switch (control->type) { case WM8994: if (wm8994->micdet_irq) { @@ -4313,6 +4307,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } wm_hubs_add_analogue_routes(codec, 0, 0); + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, + wm_hubs_dcs_done, "DC servo done", + &wm8994->hubs); + if (ret == 0) + wm8994->hubs.dcs_done_irq = true; snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); switch (control->type) { -- cgit v0.10.2 From 1769267bb01303ac73b48535454461819ef1dcc2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:48 -0200 Subject: ASoC: wm8776: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 942d58e..ef82467 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -532,7 +532,7 @@ static struct spi_driver wm8776_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8776_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -584,7 +584,7 @@ static struct i2c_driver wm8776_i2c_driver = { static int __init wm8776_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8776_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8776 I2C driver: %d\n", @@ -604,7 +604,7 @@ module_init(wm8776_modinit); static void __exit wm8776_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8776_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From f3f9a60f7947b6bd2f970d5680dd3df624405027 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:49 -0200 Subject: ASoC: wm8804: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..9bc8206 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -739,7 +739,7 @@ static struct spi_driver wm8804_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8804_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -791,7 +791,7 @@ static int __init wm8804_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8804_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", @@ -811,7 +811,7 @@ module_init(wm8804_modinit); static void __exit wm8804_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8804_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From f25cf34969823ab7197ce9ff2521c33f0141075b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:50 -0200 Subject: ASoC: wm8900: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 734209e..e98bc70 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1288,7 +1288,7 @@ static struct spi_driver wm8900_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8900_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1338,7 +1338,7 @@ static struct i2c_driver wm8900_i2c_driver = { static int __init wm8900_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8900_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8900 I2C driver: %d\n", @@ -1358,7 +1358,7 @@ module_init(wm8900_modinit); static void __exit wm8900_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8900_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 50c9697320434d1489d087cbf38f7907a9894609 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:52 -0200 Subject: ASoC: wm8985: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 18f2bab..271b517 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1148,7 +1148,7 @@ static struct spi_driver wm8985_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8985_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1201,7 +1201,7 @@ static int __init wm8985_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8985_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8985 I2C driver: %d\n", @@ -1221,7 +1221,7 @@ module_init(wm8985_modinit); static void __exit wm8985_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8985_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 63587116811bd23d22693b50447a2a356602e70b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:53 -0200 Subject: ASoC: wm8988: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 39b9acc..a55e1c2 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -912,7 +912,7 @@ static struct spi_driver wm8988_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8988_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -964,7 +964,7 @@ static struct i2c_driver wm8988_i2c_driver = { static int __init wm8988_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8988_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8988 I2C driver: %d\n", @@ -984,7 +984,7 @@ module_init(wm8988_modinit); static void __exit wm8988_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8988_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v0.10.2 From 3f3002692ce8fa1e9b257183ea1a36baacfdcfcf Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:48:32 -0200 Subject: AsoC: wm9081: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 630b3d7..0982c1d 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1326,7 +1326,7 @@ static const struct regmap_config wm9081_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- cgit v0.10.2 From 9a199b8e9933edf83585bac2c9030870e014381b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:48:32 -0200 Subject: ASoC: wm9081: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 630b3d7..0982c1d 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1326,7 +1326,7 @@ static const struct regmap_config wm9081_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- cgit v0.10.2 From 7ae10ed2ee757f2ce19188e540eaa44f337c7cd2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:39 -0200 Subject: ASoC: cs42l52: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8b427c9..19ee10b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -50,7 +50,7 @@ struct cs42l52_private { u8 mclksel; u32 mclk; u8 flags; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; @@ -953,7 +953,7 @@ static int cs42l52_resume(struct snd_soc_codec *codec) return 0; } -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 -- cgit v0.10.2 From a3d36bc2aba531328f7311ef57dec7687283ec57 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Wed, 13 Nov 2013 16:05:40 -0600 Subject: ASoC: cs42l52: Reorganize MICA/B Config and Select This patch reworks the MICA an MICB config for single-ended or differential and the selection of which MIC for the single config Signed-off-by: Brian Austin Signed-off-by: Mark Brown diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index 7c2be4a..daa91f3 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -22,12 +22,6 @@ struct cs42l52_platform_data { /* MICB mode selection 0=Single 1=Differential */ unsigned int micb_cfg; - /* MICA Select 0=MIC1A 1=MIC2A */ - unsigned int mica_sel; - - /* MICB Select 0=MIC2A 1=MIC2B */ - unsigned int micb_sel; - /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 19ee10b..1801063 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -233,7 +233,7 @@ static const struct soc_enum mic_bias_level_enum = SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); -static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; +static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" }; static const struct soc_enum mica_enum = SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, @@ -243,12 +243,6 @@ static const struct soc_enum micb_enum = SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); -static const struct snd_kcontrol_new mica_mux = - SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); - -static const struct snd_kcontrol_new micb_mux = - SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); - static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; static const struct soc_enum digital_output_mux_enum = @@ -425,6 +419,9 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + SOC_ENUM("MICA Select", mica_enum), + SOC_ENUM("MICB Select", micb_enum), + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), @@ -550,9 +547,6 @@ static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), - SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), - SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), @@ -1239,17 +1233,6 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, cs42l52->pdata.micb_cfg << CS42L52_MIC_CTL_TYPE_SHIFT); - if (cs42l52->pdata.mica_sel) - regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.mica_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - if (cs42l52->pdata.micb_sel) - regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.micb_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - if (cs42l52->pdata.chgfreq) regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP, CS42L52_CHARGE_PUMP_MASK, -- cgit v0.10.2 From 44b2ed54036ecec36ad27adf356f0274a72e5f05 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 14 Nov 2013 11:46:11 -0600 Subject: ASoC: cs42l52: Make MICA/B mixer dependent on mic config MICA/B Single-Ended input selection depends on mica/b config so lets make the mixer controls for them only show for selected mic's Signed-off-by: Brian Austin Signed-off-by: Mark Brown diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index daa91f3..bbabf84 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -16,11 +16,11 @@ struct cs42l52_platform_data { /* MICBIAS Level. Check datasheet Pg48 */ unsigned int micbias_lvl; - /* MICA mode selection 0=Single 1=Differential */ - unsigned int mica_cfg; + /* MICA mode selection Differential or Single-ended */ + bool mica_diff_cfg; - /* MICB mode selection 0=Single 1=Differential */ - unsigned int micb_cfg; + /* MICB mode selection Differential or Single-ended */ + bool micb_diff_cfg; /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 1801063..78d2dd6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -419,9 +419,6 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), - SOC_ENUM("MICA Select", mica_enum), - SOC_ENUM("MICB Select", micb_enum), - SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), @@ -528,6 +525,30 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { }; +static const struct snd_kcontrol_new cs42l52_mica_controls[] = { + SOC_ENUM("MICA Select", mica_enum), +}; + +static const struct snd_kcontrol_new cs42l52_micb_controls[] = { + SOC_ENUM("MICB Select", micb_enum), +}; + +static int cs42l52_add_mic_controls(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + struct cs42l52_platform_data *pdata = &cs42l52->pdata; + + if (!pdata->mica_diff_cfg) + snd_soc_add_codec_controls(codec, cs42l52_mica_controls, + ARRAY_SIZE(cs42l52_mica_controls)); + + if (!pdata->micb_diff_cfg) + snd_soc_add_codec_controls(codec, cs42l52_micb_controls, + ARRAY_SIZE(cs42l52_micb_controls)); + + return 0; +} + static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AIN1L"), @@ -1104,6 +1125,8 @@ static int cs42l52_probe(struct snd_soc_codec *codec) } regcache_cache_only(cs42l52->regmap, true); + cs42l52_add_mic_controls(codec); + cs42l52_init_beep(codec); cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1221,16 +1244,16 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, reg & 0xFF); /* Set Platform Data */ - if (cs42l52->pdata.mica_cfg) + if (cs42l52->pdata.mica_diff_cfg) regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.mica_cfg << + cs42l52->pdata.mica_diff_cfg << CS42L52_MIC_CTL_TYPE_SHIFT); - if (cs42l52->pdata.micb_cfg) + if (cs42l52->pdata.micb_diff_cfg) regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.micb_cfg << + cs42l52->pdata.micb_diff_cfg << CS42L52_MIC_CTL_TYPE_SHIFT); if (cs42l52->pdata.chgfreq) -- cgit v0.10.2 From 32a8fda90f52f547719b16dec07cce4abbd093e0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 27 Nov 2013 12:51:17 +0000 Subject: mfd: wm5110: Give new AIF2 registers defaults and mark as readable The registers associated with the new channels on AIF2 were accidentally missing defaults and not marked as readable this patch fixes this. Signed-off-by: Charles Keepax Acked-by: Lee Jones Signed-off-by: Mark Brown diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index d433e28..338cfbe 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -885,6 +885,38 @@ static const struct reg_default wm5110_reg_default[] = { { 0x0000074D, 0x0080 }, /* R1869 - AIF2TX2MIX Input 3 Volume */ { 0x0000074E, 0x0000 }, /* R1870 - AIF2TX2MIX Input 4 Source */ { 0x0000074F, 0x0080 }, /* R1871 - AIF2TX2MIX Input 4 Volume */ + { 0x00000750, 0x0000 }, /* R1872 - AIF2TX3MIX Input 1 Source */ + { 0x00000751, 0x0080 }, /* R1873 - AIF2TX3MIX Input 1 Volume */ + { 0x00000752, 0x0000 }, /* R1874 - AIF2TX3MIX Input 2 Source */ + { 0x00000753, 0x0080 }, /* R1875 - AIF2TX3MIX Input 2 Volume */ + { 0x00000754, 0x0000 }, /* R1876 - AIF2TX3MIX Input 3 Source */ + { 0x00000755, 0x0080 }, /* R1877 - AIF2TX3MIX Input 3 Volume */ + { 0x00000756, 0x0000 }, /* R1878 - AIF2TX3MIX Input 4 Source */ + { 0x00000757, 0x0080 }, /* R1879 - AIF2TX3MIX Input 4 Volume */ + { 0x00000758, 0x0000 }, /* R1880 - AIF2TX4MIX Input 1 Source */ + { 0x00000759, 0x0080 }, /* R1881 - AIF2TX4MIX Input 1 Volume */ + { 0x0000075A, 0x0000 }, /* R1882 - AIF2TX4MIX Input 2 Source */ + { 0x0000075B, 0x0080 }, /* R1883 - AIF2TX4MIX Input 2 Volume */ + { 0x0000075C, 0x0000 }, /* R1884 - AIF2TX4MIX Input 3 Source */ + { 0x0000075D, 0x0080 }, /* R1885 - AIF2TX4MIX Input 3 Volume */ + { 0x0000075E, 0x0000 }, /* R1886 - AIF2TX4MIX Input 4 Source */ + { 0x0000075F, 0x0080 }, /* R1887 - AIF2TX4MIX Input 4 Volume */ + { 0x00000760, 0x0000 }, /* R1888 - AIF2TX5MIX Input 1 Source */ + { 0x00000761, 0x0080 }, /* R1889 - AIF2TX5MIX Input 1 Volume */ + { 0x00000762, 0x0000 }, /* R1890 - AIF2TX5MIX Input 2 Source */ + { 0x00000763, 0x0080 }, /* R1891 - AIF2TX5MIX Input 2 Volume */ + { 0x00000764, 0x0000 }, /* R1892 - AIF2TX5MIX Input 3 Source */ + { 0x00000765, 0x0080 }, /* R1893 - AIF2TX5MIX Input 3 Volume */ + { 0x00000766, 0x0000 }, /* R1894 - AIF2TX5MIX Input 4 Source */ + { 0x00000767, 0x0080 }, /* R1895 - AIF2TX5MIX Input 4 Volume */ + { 0x00000768, 0x0000 }, /* R1896 - AIF2TX6MIX Input 1 Source */ + { 0x00000769, 0x0080 }, /* R1897 - AIF2TX6MIX Input 1 Volume */ + { 0x0000076A, 0x0000 }, /* R1898 - AIF2TX6MIX Input 2 Source */ + { 0x0000076B, 0x0080 }, /* R1899 - AIF2TX6MIX Input 2 Volume */ + { 0x0000076C, 0x0000 }, /* R1900 - AIF2TX6MIX Input 3 Source */ + { 0x0000076D, 0x0080 }, /* R1901 - AIF2TX6MIX Input 3 Volume */ + { 0x0000076E, 0x0000 }, /* R1902 - AIF2TX6MIX Input 4 Source */ + { 0x0000076F, 0x0080 }, /* R1903 - AIF2TX6MIX Input 4 Volume */ { 0x00000780, 0x0000 }, /* R1920 - AIF3TX1MIX Input 1 Source */ { 0x00000781, 0x0080 }, /* R1921 - AIF3TX1MIX Input 1 Volume */ { 0x00000782, 0x0000 }, /* R1922 - AIF3TX1MIX Input 2 Source */ @@ -1883,6 +1915,38 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_AIF2TX2MIX_INPUT_3_VOLUME: case ARIZONA_AIF2TX2MIX_INPUT_4_SOURCE: case ARIZONA_AIF2TX2MIX_INPUT_4_VOLUME: + case ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE: + case ARIZONA_AIF2TX3MIX_INPUT_1_VOLUME: + case ARIZONA_AIF2TX3MIX_INPUT_2_SOURCE: + case ARIZONA_AIF2TX3MIX_INPUT_2_VOLUME: + case ARIZONA_AIF2TX3MIX_INPUT_3_SOURCE: + case ARIZONA_AIF2TX3MIX_INPUT_3_VOLUME: + case ARIZONA_AIF2TX3MIX_INPUT_4_SOURCE: + case ARIZONA_AIF2TX3MIX_INPUT_4_VOLUME: + case ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE: + case ARIZONA_AIF2TX4MIX_INPUT_1_VOLUME: + case ARIZONA_AIF2TX4MIX_INPUT_2_SOURCE: + case ARIZONA_AIF2TX4MIX_INPUT_2_VOLUME: + case ARIZONA_AIF2TX4MIX_INPUT_3_SOURCE: + case ARIZONA_AIF2TX4MIX_INPUT_3_VOLUME: + case ARIZONA_AIF2TX4MIX_INPUT_4_SOURCE: + case ARIZONA_AIF2TX4MIX_INPUT_4_VOLUME: + case ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE: + case ARIZONA_AIF2TX5MIX_INPUT_1_VOLUME: + case ARIZONA_AIF2TX5MIX_INPUT_2_SOURCE: + case ARIZONA_AIF2TX5MIX_INPUT_2_VOLUME: + case ARIZONA_AIF2TX5MIX_INPUT_3_SOURCE: + case ARIZONA_AIF2TX5MIX_INPUT_3_VOLUME: + case ARIZONA_AIF2TX5MIX_INPUT_4_SOURCE: + case ARIZONA_AIF2TX5MIX_INPUT_4_VOLUME: + case ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE: + case ARIZONA_AIF2TX6MIX_INPUT_1_VOLUME: + case ARIZONA_AIF2TX6MIX_INPUT_2_SOURCE: + case ARIZONA_AIF2TX6MIX_INPUT_2_VOLUME: + case ARIZONA_AIF2TX6MIX_INPUT_3_SOURCE: + case ARIZONA_AIF2TX6MIX_INPUT_3_VOLUME: + case ARIZONA_AIF2TX6MIX_INPUT_4_SOURCE: + case ARIZONA_AIF2TX6MIX_INPUT_4_VOLUME: case ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE: case ARIZONA_AIF3TX1MIX_INPUT_1_VOLUME: case ARIZONA_AIF3TX1MIX_INPUT_2_SOURCE: -- cgit v0.10.2 From 391fc59db87615e07e8a6ab5fbffe3cc04f2b19c Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 15 Nov 2013 09:35:33 -0600 Subject: ASoC: cs42l52: Add devicetree support for CS42L52 This patch adds device tree support for the CS42L52 Codec Signed-off-by: Brian Austin Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 78d2dd6..4a47a63 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,7 +17,7 @@ #include #include #include -#include +#include #include #include #include @@ -1193,6 +1193,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), GFP_KERNEL); @@ -1206,9 +1207,39 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } - - if (pdata) + if (pdata) { + cs42l52->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l52_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_bool(i2c_client->dev.of_node, + "cirrus,mica-differential-cfg")) + pdata->mica_diff_cfg = true; + + if (of_property_read_bool(i2c_client->dev.of_node, + "cirrus,micb-differential-cfg")) + pdata->micb_diff_cfg = true; + + if (of_property_read_u32(i2c_client->dev.of_node, + "cirrus,micbias-lvl", &val32) >= 0) + pdata->micbias_lvl = val32; + + if (of_property_read_u32(i2c_client->dev.of_node, + "cirrus,chgfreq-divisor", &val32) >= 0) + pdata->chgfreq_divisor = val32; + + pdata->reset_gpio = + of_get_named_gpio(i2c_client->dev.of_node, + "cirrus,reset-gpio", 0); + } cs42l52->pdata = *pdata; + } if (cs42l52->pdata.reset_gpio) { ret = gpio_request_one(cs42l52->pdata.reset_gpio, @@ -1280,6 +1311,13 @@ static int cs42l52_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l52_of_match[] = { + { .compatible = "cirrus,cs42l52", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l52_of_match); + + static const struct i2c_device_id cs42l52_id[] = { { "cs42l52", 0 }, { } @@ -1290,6 +1328,7 @@ static struct i2c_driver cs42l52_i2c_driver = { .driver = { .name = "cs42l52", .owner = THIS_MODULE, + .of_match_table = cs42l52_of_match, }, .id_table = cs42l52_id, .probe = cs42l52_i2c_probe, -- cgit v0.10.2 From 3a85ca9d8a06c873b7a5fb24319572926fa20e10 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 15 Nov 2013 09:35:34 -0600 Subject: ASoC: dt: binding: sound cs42l52 driver Signed-off-by: Brian Austin Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/cs42l52.txt b/Documentation/devicetree/bindings/sound/cs42l52.txt new file mode 100644 index 0000000..bc03c93 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l52.txt @@ -0,0 +1,46 @@ +CS42L52 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l52" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - cirrus,reset-gpio : GPIO controller's phandle and the number + of the GPIO used to reset the codec. + + - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency. + Allowable values of 0x00 through 0x0F. These are raw values written to the + register, not the actual frequency. The frequency is determined by the following. + Frequency = (64xFs)/(N+2) + N = chgfreq_val + Fs = Sample Rate (variable) + + - cirrus,mica-differential-cfg : boolean, If present, then the MICA input is configured + as a differential input. If not present then the MICA input is configured as + Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input. + + - cirrus,micb-differential-cfg : boolean, If present, then the MICB input is configured + as a differential input. If not present then the MICB input is configured as + Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input. + + - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin + 0 = 0.5 x VA + 1 = 0.6 x VA + 2 = 0.7 x VA + 3 = 0.8 x VA + 4 = 0.83 x VA + 5 = 0.91 x VA + +Example: + +codec: codec@4a { + compatible = "cirrus,cs42l52"; + reg = <0x4a>; + reset-gpio = <&gpio 10 0>; + cirrus,chgfreq-divisor = <0x05>; + cirrus.mica-differential-cfg; + cirrus,micbias-lvl = <5>; +}; -- cgit v0.10.2 From d71b3ef44f9e5cfda2499768f6420b784845af06 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:36 +0100 Subject: ASoC: spear: Use devm_snd_dmaengine_pcm_register Makes the code slightly shorter. Signed-off-by: Lars-Peter Clausen Acked-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 4707f2b..9a02141 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -49,18 +49,12 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { static int spear_soc_platform_probe(struct platform_device *pdev) { - return snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(&pdev->dev, &spear_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } -static int spear_soc_platform_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - return 0; -} - static struct platform_driver spear_pcm_driver = { .driver = { .name = "spear-pcm-audio", @@ -68,7 +62,6 @@ static struct platform_driver spear_pcm_driver = { }, .probe = spear_soc_platform_probe, - .remove = spear_soc_platform_remove, }; module_platform_driver(spear_pcm_driver); -- cgit v0.10.2 From d733dc0828cfb230171ae7420a6e8c344ec8473a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 28 Nov 2013 16:37:51 +0000 Subject: ASoC: wm_adsp: Stop region iteration when the desired region is found When locating the memory region relating to a coefficient block written through a bin file we keep processing the list of regions even after we have found the region we require. This patch adds a break, so we don't process redundant list items. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9..b42f9af 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1286,6 +1286,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) reg = wm_adsp_region_to_reg(mem, reg); reg += offset; + break; } } -- cgit v0.10.2 From 17b9a2b78586c42916a2bfc55ea6c0ef962b2c1e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:42:49 -0800 Subject: ASoC: rcar: remove unused register settings AUDIO_CLK_SEL4/5 are not used Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 61212ee..b94d4ce 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -208,9 +208,6 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index b5ac3a2..63a9d70 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -49,9 +49,6 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, - RSND_REG_AUDIO_CLK_SEL3, - RSND_REG_AUDIO_CLK_SEL4, - RSND_REG_AUDIO_CLK_SEL5, /* SSI */ RSND_REG_SSICR, -- cgit v0.10.2 From c1e6cc5e577d1d446a645aea02d28a924f20b834 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:01 -0800 Subject: ASoC: rcar: separate regmap init common field The repmap initialization difference between Gen1/Gen2 is only register offset. This patch separates rsnd_gen1_regmap_init() into common part and Gen1 specific part. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index b94d4ce..0ebea44 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -115,6 +115,33 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, mask, data); } +static int rsnd_gen_regmap_init(struct rsnd_priv *priv, + struct rsnd_gen *gen, + struct reg_field *regf) +{ + int i; + struct device *dev = rsnd_priv_to_dev(priv); + struct regmap_config regc; + + memset(®c, 0, sizeof(regc)); + regc.reg_bits = 32; + regc.val_bits = 32; + + gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); + if (IS_ERR(gen->regmap)) { + dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); + return PTR_ERR(gen->regmap); + } + + for (i = 0; i < RSND_REG_MAX; i++) { + gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); + if (IS_ERR(gen->regs[i])) + return PTR_ERR(gen->regs[i]); + + } + + return 0; +} /* * Gen2 * will be filled in the future @@ -189,9 +216,6 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { - int i; - struct device *dev = rsnd_priv_to_dev(priv); - struct regmap_config regc; struct reg_field regf[RSND_REG_MAX] = { RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00), RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), @@ -216,24 +240,7 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), }; - memset(®c, 0, sizeof(regc)); - regc.reg_bits = 32; - regc.val_bits = 32; - - gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); - if (IS_ERR(gen->regmap)) { - dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); - return PTR_ERR(gen->regmap); - } - - for (i = 0; i < RSND_REG_MAX; i++) { - gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); - if (IS_ERR(gen->regs[i])) - return PTR_ERR(gen->regs[i]); - - } - - return 0; + return rsnd_gen_regmap_init(priv, gen, regf); } static int rsnd_gen1_probe(struct platform_device *pdev, -- cgit v0.10.2 From 42ee5d22e3d2550a49bc5d3e6f19c92da9a19446 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:13 -0800 Subject: ASoC: rcar: add rsnd_is_accessible_reg() Current rcar driver is supporting Gen1, and Gen2 will be supported soon. Then, some registers are used from Gen1 only, or from Gen2 only. To avoid NULL pointer access, this patch adds register accessible check function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0ebea44..970439d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -86,12 +86,28 @@ static struct regmap_bus rsnd_regmap_bus = { .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; +static int rsnd_is_accessible_reg(struct rsnd_priv *priv, + struct rsnd_gen *gen, enum rsnd_reg reg) +{ + if (!gen->regs[reg]) { + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "unsupported register access %x\n", reg); + return 0; + } + + return 1; +} + u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return 0; + regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); return val; @@ -103,6 +119,9 @@ void rsnd_write(struct rsnd_priv *priv, { struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return; + regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); } @@ -111,6 +130,9 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, { struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return; + regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), mask, data); } @@ -134,6 +156,10 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, } for (i = 0; i < RSND_REG_MAX; i++) { + gen->regs[i] = NULL; + if (!regf[i].reg) + continue; + gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); if (IS_ERR(gen->regs[i])) return PTR_ERR(gen->regs[i]); -- cgit v0.10.2 From 994a9df1e3ea3244e94309e4f893e9a5121116c9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:23 -0800 Subject: ASoC: rcar: remove .path_init/exit from rsnd_gen_ops rsnd_gen_ops has .path_init/exit callback function which cares SRU/SSI (if Gen1) SCU/SSIU/SSI (if Gen2) path settings. But, the differences between Gen1/Gen2 are cared in ssi.c/scu.c, and the path itself is same in Gen1/Gen2. This patch removes .path_init/exit callback. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 970439d..4f2eaa3 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -16,12 +16,6 @@ struct rsnd_gen_ops { struct rsnd_priv *priv); void (*remove)(struct platform_device *pdev, struct rsnd_priv *priv); - int (*path_init)(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io); - int (*path_exit)(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io); }; struct rsnd_gen { @@ -168,17 +162,10 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, return 0; } -/* - * Gen2 - * will be filled in the future - */ -/* - * Gen1 - */ -static int rsnd_gen1_path_init(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) +int rsnd_gen_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) { struct rsnd_mod *mod; int ret; @@ -216,9 +203,9 @@ static int rsnd_gen1_path_init(struct rsnd_priv *priv, return ret; } -static int rsnd_gen1_path_exit(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) +int rsnd_gen_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) { struct rsnd_mod *mod, *n; int ret = 0; @@ -232,6 +219,15 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, return ret; } +/* + * Gen2 + * will be filled in the future + */ + +/* + * Gen1 + */ + /* single address mapping */ #define RSND_GEN1_S_REG(gen, reg, id, offset) \ RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9) @@ -319,31 +315,11 @@ static void rsnd_gen1_remove(struct platform_device *pdev, static struct rsnd_gen_ops rsnd_gen1_ops = { .probe = rsnd_gen1_probe, .remove = rsnd_gen1_remove, - .path_init = rsnd_gen1_path_init, - .path_exit = rsnd_gen1_path_exit, }; /* * Gen */ -int rsnd_gen_path_init(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - - return gen->ops->path_init(priv, rdai, io); -} - -int rsnd_gen_path_exit(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - - return gen->ops->path_exit(priv, rdai, io); -} - int rsnd_gen_probe(struct platform_device *pdev, struct rcar_snd_info *info, struct rsnd_priv *priv) -- cgit v0.10.2 From 531eaf491e25ce215bbcf434e23e77f53fc98171 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:34 -0800 Subject: ASoC: rcar: remove rcar_gen_ops Current rcar driver gen.c is using rcar_gen_ops which was made with the assumption that Gen1 and Gen2 need different behavior. but it was not needed. This patch removes unnecessary complex method. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 4f2eaa3..a29e36e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -10,14 +10,6 @@ */ #include "rsnd.h" -struct rsnd_gen_ops { - int (*probe)(struct platform_device *pdev, - struct rcar_snd_info *info, - struct rsnd_priv *priv); - void (*remove)(struct platform_device *pdev, - struct rsnd_priv *priv); -}; - struct rsnd_gen { void __iomem *base[RSND_BASE_MAX]; @@ -307,16 +299,6 @@ static int rsnd_gen1_probe(struct platform_device *pdev, } -static void rsnd_gen1_remove(struct platform_device *pdev, - struct rsnd_priv *priv) -{ -} - -static struct rsnd_gen_ops rsnd_gen1_ops = { - .probe = rsnd_gen1_probe, - .remove = rsnd_gen1_remove, -}; - /* * Gen */ @@ -326,6 +308,7 @@ int rsnd_gen_probe(struct platform_device *pdev, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen; + int ret; gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); if (!gen) { @@ -333,23 +316,19 @@ int rsnd_gen_probe(struct platform_device *pdev, return -ENOMEM; } - if (rsnd_is_gen1(priv)) - gen->ops = &rsnd_gen1_ops; + priv->gen = gen; - if (!gen->ops) { + if (rsnd_is_gen1(priv)) { + ret = rsnd_gen1_probe(pdev, info, priv); + } else { dev_err(dev, "unknown generation R-Car sound device\n"); return -ENODEV; } - priv->gen = gen; - - return gen->ops->probe(pdev, info, priv); + return ret; } void rsnd_gen_remove(struct platform_device *pdev, struct rsnd_priv *priv) { - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - - gen->ops->remove(pdev, priv); } -- cgit v0.10.2 From 507d466c733e28d132a7be87040a3da126df7947 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:45 -0800 Subject: ASoC: rcar: add Gen2 sound support This patch adds Gen2 sound support for Renesas R-Car. But, it is supporting PIO transfer only at this point Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index a29e36e..bf066f7 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -213,9 +213,88 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, /* * Gen2 - * will be filled in the future */ +/* single address mapping */ +#define RSND_GEN2_S_REG(gen, reg, id, offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9) + +/* multi address mapping */ +#define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9) + +static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) +{ + struct reg_field regf[RSND_REG_MAX] = { + RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800), + RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804), + /* FIXME: it needs SSI_MODE2/3 in the future */ + RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80), + + RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00), + RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04), + RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08), + RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), + RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), + RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14), + + RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSITDR, 0x08, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), + }; + + return rsnd_gen_regmap_init(priv, gen, regf); +} + +static int rsnd_gen2_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + struct resource *scu_res; + struct resource *adg_res; + struct resource *ssiu_res; + struct resource *ssi_res; + int ret; + + /* + * map address + */ + scu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SCU); + adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_ADG); + ssiu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSIU); + ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI); + + gen->base[RSND_GEN2_SCU] = devm_ioremap_resource(dev, scu_res); + gen->base[RSND_GEN2_ADG] = devm_ioremap_resource(dev, adg_res); + gen->base[RSND_GEN2_SSIU] = devm_ioremap_resource(dev, ssiu_res); + gen->base[RSND_GEN2_SSI] = devm_ioremap_resource(dev, ssi_res); + if (IS_ERR(gen->base[RSND_GEN2_SCU]) || + IS_ERR(gen->base[RSND_GEN2_ADG]) || + IS_ERR(gen->base[RSND_GEN2_SSIU]) || + IS_ERR(gen->base[RSND_GEN2_SSI])) + return -ENODEV; + + ret = rsnd_gen2_regmap_init(priv, gen); + if (ret < 0) + return ret; + + dev_dbg(dev, "Gen2 device probed\n"); + dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start, + gen->base[RSND_GEN2_SCU]); + dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + gen->base[RSND_GEN2_ADG]); + dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + gen->base[RSND_GEN2_SSIU]); + dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + gen->base[RSND_GEN2_SSI]); + + return 0; +} + /* * Gen1 */ @@ -318,12 +397,14 @@ int rsnd_gen_probe(struct platform_device *pdev, priv->gen = gen; - if (rsnd_is_gen1(priv)) { + ret = -ENODEV; + if (rsnd_is_gen1(priv)) ret = rsnd_gen1_probe(pdev, info, priv); - } else { + else if (rsnd_is_gen2(priv)) + ret = rsnd_gen2_probe(pdev, info, priv); + + if (ret < 0) dev_err(dev, "unknown generation R-Car sound device\n"); - return -ENODEV; - } return ret; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 63a9d70..bff7b9e 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -31,7 +31,7 @@ * see gen1/gen2 for detail */ enum rsnd_reg { - /* SRU/SCU */ + /* SRU/SCU/SSIU */ RSND_REG_SRC_ROUTE_SEL, RSND_REG_SRC_TMG_SEL0, RSND_REG_SRC_TMG_SEL1, @@ -41,6 +41,7 @@ enum rsnd_reg { RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, RSND_REG_BUSIF_ADINR, + RSND_REG_INT_ENABLE, /* ADG */ RSND_REG_BRRA, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index b71cf9d..477465f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -457,6 +457,9 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; + /* enable PIO interrupt */ + rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000); + rsnd_ssi_hw_start(ssi, rdai, io); dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); -- cgit v0.10.2 From 69ae8489076fa0fa98609155434f3c286c7364a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Nov 2013 11:42:18 +0000 Subject: ASoC: cs42l52: Fix build Reported-by: Stephen Rothwell Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4a47a63..0bac6d5 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1232,7 +1232,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, if (of_property_read_u32(i2c_client->dev.of_node, "cirrus,chgfreq-divisor", &val32) >= 0) - pdata->chgfreq_divisor = val32; + pdata->chgfreq = val32; pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, -- cgit v0.10.2 From 4ded61eb3ea87e9c563e09662be3ed5e942ff2a2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 2 Dec 2013 00:41:24 -0200 Subject: ASoC: imx-spdif: Remove error message upon devm_kzalloc() failure No need to have a specific OOM message, since there is generic MM out of memory message in place. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 8499d52..980dd1f 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -35,7 +35,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { - dev_err(&pdev->dev, "failed to allocate memory\n"); ret = -ENOMEM; goto end; } -- cgit v0.10.2 From 3990c516de66af940c5c366a81531787aefe81ae Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:33 +0100 Subject: ASoC: bcm2835-i2s: Use devm_snd_dmaengine_pcm_register() Makes the code slightly shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index f49b007..2685fe4 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -848,7 +848,7 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) return ret; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); return ret; @@ -862,15 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; -static int bcm2835_i2s_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - return 0; -} - static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, - .remove = bcm2835_i2s_remove, .driver = { .name = "bcm2835-i2s", .owner = THIS_MODULE, -- cgit v0.10.2 From 237eeb1c044fdd0f406a8484ee31884e34b9dfc5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:34 +0100 Subject: ASoC: ep93xx: Use devm_snd_dmaengine_pcm_register() Makes the code slightly shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index cfe517e..fdb8b8f 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -78,19 +78,13 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { static int ep93xx_soc_platform_probe(struct platform_device *pdev) { - return snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(&pdev->dev, &ep93xx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } -static int ep93xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - return 0; -} - static struct platform_driver ep93xx_pcm_driver = { .driver = { .name = "ep93xx-pcm-audio", @@ -98,7 +92,6 @@ static struct platform_driver ep93xx_pcm_driver = { }, .probe = ep93xx_soc_platform_probe, - .remove = ep93xx_soc_platform_remove, }; module_platform_driver(ep93xx_pcm_driver); -- cgit v0.10.2 From 7e6d18ac7ea1372b462778ff7c416ceaabe71b66 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:35 +0100 Subject: ASoC: fsl: Use devm_snd_dmaengine_pcm_register() Makes the code shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 55193a5..4d075f1 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1181,13 +1181,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } -static int fsl_spdif_remove(struct platform_device *pdev) -{ - imx_pcm_dma_exit(pdev); - - return 0; -} - static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx35-spdif", }, {} @@ -1201,7 +1194,6 @@ static struct platform_driver fsl_spdif_driver = { .of_match_table = fsl_spdif_dt_ids, }, .probe = fsl_spdif_probe, - .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fb8f52a..3df0318 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1109,8 +1109,6 @@ done: return 0; error_dai: - if (ssi_private->ssi_on_imx) - imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); error_dev: @@ -1132,8 +1130,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) - imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index aee2307..c5e47f8 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -61,16 +61,11 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { int imx_pcm_dma_init(struct platform_device *pdev) { - return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config, + return devm_snd_dmaengine_pcm_register(&pdev->dev, + &imx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -void imx_pcm_dma_exit(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); -} -EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); - MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 5d5b733..c79cb27 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -40,16 +40,11 @@ struct imx_pcm_fiq_params { #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); -void imx_pcm_dma_exit(struct platform_device *pdev); #else static inline int imx_pcm_dma_init(struct platform_device *pdev) { return -ENODEV; } - -static inline void imx_pcm_dma_exit(struct platform_device *pdev) -{ -} #endif #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f5f248c..cc7376f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -624,9 +624,6 @@ static int imx_ssi_remove(struct platform_device *pdev) { struct imx_ssi *ssi = platform_get_drvdata(pdev); - if (!ssi->dma_init) - imx_pcm_dma_exit(pdev); - if (!ssi->fiq_init) imx_pcm_fiq_exit(pdev); -- cgit v0.10.2 From 2650bc4f6d0c36f1219d2070485cc2980a88fab3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:37 +0100 Subject: ASoC: mxs: Use devm_snd_dmaengine_pcm_register() Makes the code shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b16abbb..04a6b0d 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -56,16 +56,10 @@ static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { int mxs_pcm_platform_register(struct device *dev) { - return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, + return devm_snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); -void mxs_pcm_platform_unregister(struct device *dev) -{ - snd_dmaengine_pcm_unregister(dev); -} -EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister); - MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index bc685b6..035ea04 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -20,6 +20,5 @@ #define _MXS_PCM_H int mxs_pcm_platform_register(struct device *dev); -void mxs_pcm_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54e622a..92db74d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -804,13 +804,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return 0; } -static int mxs_saif_remove(struct platform_device *pdev) -{ - mxs_pcm_platform_unregister(&pdev->dev); - - return 0; -} - static const struct of_device_id mxs_saif_dt_ids[] = { { .compatible = "fsl,imx28-saif", }, { /* sentinel */ } @@ -819,7 +812,6 @@ MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids); static struct platform_driver mxs_saif_driver = { .probe = mxs_saif_probe, - .remove = mxs_saif_remove, .driver = { .name = "mxs-saif", -- cgit v0.10.2 From a8ca52b7911378864e6defb42be9166c248a3749 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 17:27:02 +0000 Subject: ASoC: ak4642: Convert to table based control init Improves error handling and saves code. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2f861c9..7fe1e90 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -469,9 +469,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_codec_controls(codec, ak4642_snd_controls, - ARRAY_SIZE(ak4642_snd_controls)); - ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -491,6 +488,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .reg_cache_default = ak4642_reg, /* ak4642 reg */ .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ .reg_word_size = sizeof(u8), + .controls = ak4642_snd_controls, + .num_controls = ARRAY_SIZE(ak4642_snd_controls), .dapm_widgets = ak4642_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), .dapm_routes = ak4642_intercon, @@ -505,6 +504,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { .reg_cache_default = ak4648_reg, /* ak4648 reg */ .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ .reg_word_size = sizeof(u8), + .controls = ak4642_snd_controls, + .num_controls = ARRAY_SIZE(ak4642_snd_controls), .dapm_widgets = ak4642_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), .dapm_routes = ak4642_intercon, -- cgit v0.10.2 From 4574cd94a717eff3021a3e187dd48846adbd21ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 18:03:49 +0000 Subject: ASoC: ak4642: Convert to direct regmap API usage This moves us towards being able to remove the ASoC level I/O code which duplicates regmap functionality. Currently the only difference between the supported devices in the driver is the regmap so we can replace the CODEC driver selections with regmap selection instead. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7fe1e90..5af2374 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -198,30 +199,30 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { /* * ak4642 register cache */ -static const u8 ak4642_reg[] = { - 0x00, 0x00, 0x01, 0x00, - 0x02, 0x00, 0x00, 0x00, - 0xe1, 0xe1, 0x18, 0x00, - 0xe1, 0x18, 0x11, 0x08, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, +static const struct reg_default ak4642_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, + { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, + { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, }; -static const u8 ak4648_reg[] = { - 0x00, 0x00, 0x01, 0x00, - 0x02, 0x00, 0x00, 0x00, - 0xe1, 0xe1, 0x18, 0x00, - 0xe1, 0x18, 0x11, 0xb8, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x88, 0x88, 0x08, +static const struct reg_default ak4648_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, + { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, + { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 }, }; static int ak4642_dai_startup(struct snd_pcm_substream *substream, @@ -454,7 +455,10 @@ static struct snd_soc_dai_driver ak4642_dai = { static int ak4642_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + regcache_sync(regmap); return 0; } @@ -463,7 +467,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -485,9 +489,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .remove = ak4642_remove, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, - .reg_cache_default = ak4642_reg, /* ak4642 reg */ - .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ - .reg_word_size = sizeof(u8), .controls = ak4642_snd_controls, .num_controls = ARRAY_SIZE(ak4642_snd_controls), .dapm_widgets = ak4642_dapm_widgets, @@ -496,20 +497,20 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; -static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { - .probe = ak4642_probe, - .remove = ak4642_remove, - .resume = ak4642_resume, - .set_bias_level = ak4642_set_bias_level, - .reg_cache_default = ak4648_reg, /* ak4648 reg */ - .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ - .reg_word_size = sizeof(u8), - .controls = ak4642_snd_controls, - .num_controls = ARRAY_SIZE(ak4642_snd_controls), - .dapm_widgets = ak4642_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), - .dapm_routes = ak4642_intercon, - .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), +static const struct regmap_config ak4642_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .reg_defaults = ak4642_reg, + .num_reg_defaults = ARRAY_SIZE(ak4642_reg), +}; + +static const struct regmap_config ak4648_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .reg_defaults = ak4648_reg, + .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; #if IS_ENABLED(CONFIG_I2C) @@ -518,26 +519,30 @@ static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct device_node *np = i2c->dev.of_node; - const struct snd_soc_codec_driver *driver; + const struct regmap_config *regmap_config = NULL; + struct regmap *regmap; - driver = NULL; if (np) { const struct of_device_id *of_id; of_id = of_match_device(ak4642_of_match, &i2c->dev); if (of_id) - driver = of_id->data; + regmap_config = of_id->data; } else { - driver = (struct snd_soc_codec_driver *)id->driver_data; + regmap_config = (const struct regmap_config *)id->driver_data; } - if (!driver) { - dev_err(&i2c->dev, "no driver\n"); + if (!regmap_config) { + dev_err(&i2c->dev, "Unknown device type\n"); return -EINVAL; } + regmap = devm_regmap_init_i2c(i2c, regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&i2c->dev, - driver, &ak4642_dai, 1); + &soc_codec_dev_ak4642, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -547,17 +552,17 @@ static int ak4642_i2c_remove(struct i2c_client *client) } static struct of_device_id ak4642_of_match[] = { - { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, - { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, - { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + { .compatible = "asahi-kasei,ak4642", .data = &ak4642_regmap}, + { .compatible = "asahi-kasei,ak4643", .data = &ak4642_regmap}, + { .compatible = "asahi-kasei,ak4648", .data = &ak4648_regmap}, {}, }; MODULE_DEVICE_TABLE(of, ak4642_of_match); static const struct i2c_device_id ak4642_i2c_id[] = { - { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, - { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, - { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 }, + { "ak4642", (kernel_ulong_t)&ak4642_regmap }, + { "ak4643", (kernel_ulong_t)&ak4642_regmap }, + { "ak4648", (kernel_ulong_t)&ak4648_regmap }, { } }; MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); -- cgit v0.10.2 From 3621dbbc27ff347f2e4476013054bab18ebd906c Mon Sep 17 00:00:00 2001 From: Oskar Schirmer Date: Sat, 16 Nov 2013 07:52:25 +0000 Subject: ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation Unbalanced calls to imx_ssi_trigger() may result in endless SSI activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the ssi->enabled counter to increase and never return to zero again, so eventually SSI_SCR_SSIEN is never disabled. As the information on whether to enable the SSI or not is contained in the two bits for TE/RE, we save all the software mirroring of hardware state here and simply use the hardware register itself to keep the state of whether someone is currently playing or capturing. This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c which I send a patch for three days ago. Astonishing enough this highly fragile scheme is used twice in parallel to serve the very same control function, synchronously: Once out of sync you are lost until reboot. Note, that these fixes wont prevent state machine distortion on alsa level to cut sound or the like. It just makes sure we have a chance to synchronise again later on. Signed-off-by: Oskar Schirmer Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index cc7376f..6336757 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -304,8 +304,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, scr |= SSI_SCR_RE; sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; + scr |= SSI_SCR_SSIEN; break; @@ -318,7 +317,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, scr &= ~SSI_SCR_RE; sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!(scr & (SSI_SCR_TE | SSI_SCR_RE))) scr &= ~SSI_SCR_SSIEN; break; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 560c40f..be65623 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -213,7 +213,6 @@ struct imx_ssi { int fiq_init; int dma_init; - int enabled; }; #endif /* _IMX_SSI_H */ -- cgit v0.10.2 From 2f54d2a1cf7e62f56b1b0bcf44bd704f65359f38 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 17:24:55 +0000 Subject: ASoC: ak4642: Convert to module_i2c_driver() The device does not support anything other than I2C (at least with the current driver) so save code. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5af2374..1f646c6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -513,7 +513,6 @@ static const struct regmap_config ak4648_regmap = { .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; -#if IS_ENABLED(CONFIG_I2C) static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -577,27 +576,8 @@ static struct i2c_driver ak4642_i2c_driver = { .remove = ak4642_i2c_remove, .id_table = ak4642_i2c_id, }; -#endif -static int __init ak4642_modinit(void) -{ - int ret = 0; -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ak4642_i2c_driver); -#endif - return ret; - -} -module_init(ak4642_modinit); - -static void __exit ak4642_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ak4642_i2c_driver); -#endif - -} -module_exit(ak4642_exit); +module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v0.10.2 From 2924a9981006ad01efb46c754689fa7d03e3eb4f Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 2 Dec 2013 23:29:03 +0800 Subject: ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3df0318..90ff107 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -143,6 +143,7 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -354,14 +355,13 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -378,7 +378,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -508,6 +508,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -537,6 +538,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -649,14 +655,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- cgit v0.10.2 From 07a9483aaca5d3b5de8ee824ee576321d3f8b4c6 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 3 Dec 2013 18:38:07 +0800 Subject: ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits Since we introduced symmetric_channels and symmetric_samplebits, we implement these two features to fsl_ssi so as to drop some no-more-needed code and make the driver neat and clean. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 90ff107..f9f4569 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -119,8 +119,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI - * @first_stream: pointer to the stream that was opened first - * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @cpu_dai: the CPU DAI for this device @@ -132,8 +130,6 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; - struct snd_pcm_substream *first_stream; - struct snd_pcm_substream *second_stream; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -438,54 +434,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; - /* - * If this is the first stream opened, then request the IRQ - * and initialize the SSI registers. + /* First, we only do fsl_ssi_setup() when SSI is going to be active. + * Second, fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ - if (!ssi_private->first_stream) { - ssi_private->first_stream = substream; - - /* - * fsl_ssi_setup was already called by ac97_init earlier if - * the driver is in ac97 mode. - */ - if (!ssi_private->imx_ac97) - fsl_ssi_setup(ssi_private); - } else { - if (synchronous) { - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - /* - * This is the second stream open, and we're in - * synchronous mode, so we need to impose sample - * sample size constraints. This is because STCCR is - * used for playback and capture in synchronous mode, - * so there's no way to specify different word - * lengths. - * - * Note that this can cause a race condition if the - * second stream is opened before the first stream is - * fully initialized. We provide some protection by - * checking to make sure the first stream is - * initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample - * rate or size. If the second stream is opened - * before the first stream has received its final - * parameters, then the second stream may be - * constrained to the wrong sample rate or size. - */ - if (first_runtime->sample_bits) { - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); - } - } - - ssi_private->second_stream = substream; - } + if (!dai->active && !ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); return 0; } @@ -615,23 +570,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -/** - * fsl_ssi_shutdown: shutdown the SSI - * - * Shutdown the SSI if there are no other substreams open. - */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - - if (ssi_private->first_stream == substream) - ssi_private->first_stream = ssi_private->second_stream; - - ssi_private->second_stream = NULL; -} - static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); @@ -647,7 +585,6 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, }; @@ -722,7 +659,6 @@ static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { .startup = fsl_ssi_startup, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_ac97_trigger, }; @@ -947,8 +883,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } /* Are the RX and the TX clocks locked? */ - if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); -- cgit v0.10.2 From a010ff628c0953e6c914ecd09678363848617a88 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 2 Dec 2013 21:26:22 +0100 Subject: ASoC: ssm2602: Use core for applying symmetry constraints Let the core take care of applying sample rate and sample bits constraints instead of open-coding this in the driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 480074d..c6dd485 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,8 +53,6 @@ enum ssm2602_type { struct ssm2602_priv { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; struct regmap *regmap; @@ -277,11 +275,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); unsigned int iface; - if (substream == ssm2602->slave_substream) { - dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - if (srate < 0) return srate; @@ -314,33 +307,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; if (ssm2602->sysclk_constraints) { snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -351,19 +317,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; -} - - static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); @@ -530,7 +483,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, @@ -551,6 +503,8 @@ static struct snd_soc_dai_driver ssm2602_dai = { .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, .ops = &ssm2602_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, }; static int ssm2602_suspend(struct snd_soc_codec *codec) -- cgit v0.10.2 From b84c9ce809c91b3c613c967abcee90ebd6582092 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 3 Dec 2013 18:53:02 +0100 Subject: ASoC: jz4740-i2s: Use managed resources Makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 4c849a4..a0b6a85 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -432,91 +432,36 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; + struct resource *mem; int ret; - i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); - + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!i2s->mem) { - ret = -ENOENT; - goto err_free; - } - - i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), - pdev->name); - if (!i2s->mem) { - ret = -EBUSY; - goto err_free; - } - - i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); - if (!i2s->base) { - ret = -EBUSY; - goto err_release_mem_region; - } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); - i2s->phys_base = i2s->mem->start; + i2s->phys_base = mem->start; - i2s->clk_aic = clk_get(&pdev->dev, "aic"); - if (IS_ERR(i2s->clk_aic)) { - ret = PTR_ERR(i2s->clk_aic); - goto err_iounmap; - } + i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) + return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); - if (IS_ERR(i2s->clk_i2s)) { - ret = PTR_ERR(i2s->clk_i2s); - goto err_clk_put_aic; - } + i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) + return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); - ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, - &jz4740_i2s_dai, 1); - - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI\n"); - goto err_clk_put_i2s; - } - - return 0; - -err_clk_put_i2s: - clk_put(i2s->clk_i2s); -err_clk_put_aic: - clk_put(i2s->clk_aic); -err_iounmap: - iounmap(i2s->base); -err_release_mem_region: - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); -err_free: - kfree(i2s); - - return ret; -} - -static int jz4740_i2s_dev_remove(struct platform_device *pdev) -{ - struct jz4740_i2s *i2s = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - - clk_put(i2s->clk_i2s); - clk_put(i2s->clk_aic); - iounmap(i2s->base); - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - - kfree(i2s); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4740_i2s_dai, 1); } static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, - .remove = jz4740_i2s_dev_remove, .driver = { .name = "jz4740-i2s", .owner = THIS_MODULE, -- cgit v0.10.2 From 0406a40a095ca039e5f5ec63783342253c573d06 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 3 Dec 2013 18:53:03 +0100 Subject: ASoC: jz4740: Use the generic dmaengine PCM driver Now that there is a dmaengine driver for the jz4740 DMA core we can use the generic dmaengine PCM driver. This allows us to remove the custom jz4740-pcm code completely. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 5351cba..29f76af 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -1,6 +1,7 @@ config SND_JZ4740_SOC tristate "SoC Audio for Ingenic JZ4740 SoC" depends on MACH_JZ4740 && SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the JZ4740 I2S interface. You will also need to select the audio diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index a0b6a85..8f22000 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -29,9 +29,11 @@ #include #include #include +#include + +#include #include "jz4740-i2s.h" -#include "jz4740-pcm.h" #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 @@ -89,8 +91,8 @@ struct jz4740_i2s { struct clk *clk_aic; struct clk *clk_i2s; - struct jz4740_pcm_config pcm_config_playback; - struct jz4740_pcm_config pcm_config_capture; + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; }; static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, @@ -233,8 +235,6 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); - enum jz4740_dma_width dma_width; - struct jz4740_pcm_config *pcm_config; unsigned int sample_size; uint32_t ctrl; @@ -243,11 +243,9 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: sample_size = 0; - dma_width = JZ4740_DMA_WIDTH_8BIT; break; case SNDRV_PCM_FORMAT_S16: sample_size = 1; - dma_width = JZ4740_DMA_WIDTH_16BIT; break; default: return -EINVAL; @@ -260,22 +258,13 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO; else ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; - - pcm_config = &i2s->pcm_config_playback; - pcm_config->dma_config.dst_width = dma_width; - } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; - - pcm_config = &i2s->pcm_config_capture; - pcm_config->dma_config.src_width = dma_width; } jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); - snd_soc_dai_set_dma_data(dai, substream, pcm_config); - return 0; } @@ -342,25 +331,19 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai) static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) { - struct jz4740_dma_config *dma_config; + struct snd_dmaengine_dai_dma_data *dma_data; /* Playback */ - dma_config = &i2s->pcm_config_playback.dma_config; - dma_config->src_width = JZ4740_DMA_WIDTH_32BIT; - dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; - dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; - dma_config->flags = JZ4740_DMA_SRC_AUTOINC; - dma_config->mode = JZ4740_DMA_MODE_SINGLE; - i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + dma_data = &i2s->playback_dma_data; + dma_data->maxburst = 16; + dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; + dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ - dma_config = &i2s->pcm_config_capture.dma_config; - dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT; - dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; - dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; - dma_config->flags = JZ4740_DMA_DST_AUTOINC; - dma_config->mode = JZ4740_DMA_MODE_SINGLE; - i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + dma_data = &i2s->capture_dma_data; + dma_data->maxburst = 16; + dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; + dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) @@ -371,6 +354,8 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) clk_prepare_enable(i2s->clk_aic); jz4740_i2c_init_pcm_config(i2s); + snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | @@ -456,8 +441,13 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); - return devm_snd_soc_register_component(&pdev->dev, + ret = devm_snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, &jz4740_i2s_dai, 1); + if (ret) + return ret; + + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_COMPAT); } static struct platform_driver jz4740_i2s_driver = { diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c deleted file mode 100644 index 1d7ef28..0000000 --- a/sound/soc/jz4740/jz4740-pcm.c +++ /dev/null @@ -1,358 +0,0 @@ -/* - * Copyright (C) 2010, Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#include -#include -#include -#include -#include -#include - -#include - -#include -#include -#include -#include - -#include -#include "jz4740-pcm.h" - -struct jz4740_runtime_data { - unsigned long dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - - struct jz4740_dma_chan *dma; - - dma_addr_t fifo_addr; -}; - -/* identify hardware playback capabilities */ -static const struct snd_pcm_hardware jz4740_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, - - .rates = SNDRV_PCM_RATE_8000_48000, - .channels_min = 1, - .channels_max = 2, - .period_bytes_min = 16, - .period_bytes_max = 2 * PAGE_SIZE, - .periods_min = 2, - .periods_max = 128, - .buffer_bytes_max = 128 * 2 * PAGE_SIZE, - .fifo_size = 32, -}; - -static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd, - struct snd_pcm_substream *substream) -{ - unsigned long count; - - if (prtd->dma_pos == prtd->dma_end) - prtd->dma_pos = prtd->dma_start; - - if (prtd->dma_pos + prtd->dma_period > prtd->dma_end) - count = prtd->dma_end - prtd->dma_pos; - else - count = prtd->dma_period; - - jz4740_dma_disable(prtd->dma); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos); - jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr); - } else { - jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr); - jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos); - } - - jz4740_dma_set_transfer_count(prtd->dma, count); - - prtd->dma_pos += count; - - jz4740_dma_enable(prtd->dma); -} - -static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err, - void *dev_id) -{ - struct snd_pcm_substream *substream = dev_id; - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - - snd_pcm_period_elapsed(substream); - - jz4740_pcm_start_transfer(prtd, substream); -} - -static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct jz4740_pcm_config *config; - - config = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!config) - return 0; - - if (!prtd->dma) { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - prtd->dma = jz4740_dma_request(substream, "PCM Capture"); - else - prtd->dma = jz4740_dma_request(substream, "PCM Playback"); - } - - if (!prtd->dma) - return -EBUSY; - - jz4740_dma_configure(prtd->dma, &config->dma_config); - prtd->fifo_addr = config->fifo_addr; - - jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + runtime->dma_bytes; - - return 0; -} - -static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct jz4740_runtime_data *prtd = substream->runtime->private_data; - - snd_pcm_set_runtime_buffer(substream, NULL); - if (prtd->dma) { - jz4740_dma_free(prtd->dma); - prtd->dma = NULL; - } - - return 0; -} - -static int jz4740_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct jz4740_runtime_data *prtd = substream->runtime->private_data; - - if (!prtd->dma) - return -EBUSY; - - prtd->dma_pos = prtd->dma_start; - - return 0; -} - -static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - jz4740_pcm_start_transfer(prtd, substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - jz4740_dma_disable(prtd->dma); - break; - default: - break; - } - - return 0; -} - -static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - unsigned long byte_offset; - snd_pcm_uframes_t offset; - struct jz4740_dma_chan *dma = prtd->dma; - - /* prtd->dma_pos points to the end of the current transfer. So by - * subtracting prdt->dma_start we get the offset to the end of the - * current period in bytes. By subtracting the residue of the transfer - * we get the current offset in bytes. */ - byte_offset = prtd->dma_pos - prtd->dma_start; - byte_offset -= jz4740_dma_get_residue(dma); - - offset = bytes_to_frames(runtime, byte_offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int jz4740_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware); - - runtime->private_data = prtd; - - return 0; -} - -static int jz4740_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - - kfree(prtd); - - return 0; -} - -static int jz4740_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - -static struct snd_pcm_ops jz4740_pcm_ops = { - .open = jz4740_pcm_open, - .close = jz4740_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = jz4740_pcm_hw_params, - .hw_free = jz4740_pcm_hw_free, - .prepare = jz4740_pcm_prepare, - .trigger = jz4740_pcm_trigger, - .pointer = jz4740_pcm_pointer, - .mmap = jz4740_pcm_mmap, -}; - -static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = jz4740_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - buf->area = dma_alloc_noncoherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - - return 0; -} - -static void jz4740_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < SNDRV_PCM_STREAM_LAST; ++stream) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_noncoherent(pcm->card->dev, buf->bytes, buf->area, - buf->addr); - buf->area = NULL; - } -} - -static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = jz4740_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto err; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = jz4740_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto err; - } - -err: - return ret; -} - -static struct snd_soc_platform_driver jz4740_soc_platform = { - .ops = &jz4740_pcm_ops, - .pcm_new = jz4740_pcm_new, - .pcm_free = jz4740_pcm_free, -}; - -static int jz4740_pcm_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &jz4740_soc_platform); -} - -static int jz4740_pcm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver jz4740_pcm_driver = { - .probe = jz4740_pcm_probe, - .remove = jz4740_pcm_remove, - .driver = { - .name = "jz4740-pcm-audio", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(jz4740_pcm_driver); - -MODULE_AUTHOR("Lars-Peter Clausen "); -MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h deleted file mode 100644 index 1220cbb..0000000 --- a/sound/soc/jz4740/jz4740-pcm.h +++ /dev/null @@ -1,20 +0,0 @@ -/* - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _JZ4740_PCM_H -#define _JZ4740_PCM_H - -#include -#include - - -struct jz4740_pcm_config { - struct jz4740_dma_config dma_config; - phys_addr_t fifo_addr; -}; - -#endif diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 55fd6b5..82b5f37 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -73,7 +73,7 @@ static struct snd_soc_dai_link qi_lb60_dai = { .name = "jz4740", .stream_name = "jz4740", .cpu_dai_name = "jz4740-i2s", - .platform_name = "jz4740-pcm-audio", + .platform_name = "jz4740-i2s", .codec_dai_name = "jz4740-hifi", .codec_name = "jz4740-codec", .init = qi_lb60_codec_init, -- cgit v0.10.2 From 60dbb4f17417777fc56cc909c3e89f4d338e8ca8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 3 Dec 2013 22:09:33 -0800 Subject: ASoC: rcar: use devm_clk_get() instead of clk_get() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 477465f..82b04c6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -652,7 +652,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); - clk = clk_get(dev, name); + clk = devm_clk_get(dev, name); if (IS_ERR(clk)) return PTR_ERR(clk); @@ -713,7 +713,6 @@ void rsnd_ssi_remove(struct platform_device *pdev, int i; for_each_rsnd_ssi(ssi, priv, i) { - clk_put(ssi->clk); if (rsnd_ssi_dma_available(ssi)) rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); } -- cgit v0.10.2 From 1552c32547ca807f0df7d9abca54468033df8764 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 18:11:38 +0000 Subject: ASoC: adsp: Use async writes where possible This will allow a marginal speed improvement when used with a bus that supports async I/O by reducing the amount of context thrashing between writes, allowing the bus to be more fully utilised. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9..6b1c01c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1468,8 +1468,8 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) unsigned int val; int ret, count; - ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, ADSP2_SYS_ENA); + ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, ADSP2_SYS_ENA); if (ret != 0) return ret; @@ -1521,9 +1521,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, val = (val & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); if (ret != 0) { adsp_err(dsp, "Failed to set clock rate: %d\n", ret); @@ -1586,10 +1586,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; -- cgit v0.10.2 From 24f4bd57a7b24d10e52a3807f88adec79824e5d8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 Dec 2013 20:27:44 -0200 Subject: ASoC: imx-ssi: Check the return value from clk_prepare_enable() clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 6336757..df552fa 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -535,7 +535,9 @@ static int imx_ssi_probe(struct platform_device *pdev) ret); goto failed_clk; } - clk_prepare_enable(ssi->clk); + ret = clk_prepare_enable(ssi->clk); + if (ret) + goto failed_clk; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); ssi->base = devm_ioremap_resource(&pdev->dev, res); -- cgit v0.10.2 From 7637af2e17f18bfe6264d834c6edee7706a0f15c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 4 Dec 2013 15:19:27 -0700 Subject: ASoC: tegra: add tegra+MAX98090 machine driver Initially, this binding and driver only describe/support playback to headphones and speakers, and capture from the external microphone, with GPIO-based jack detection for the headphone jack only. This driver is useful for the Venice2 board. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt new file mode 100644 index 0000000..9c7c55c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -0,0 +1,51 @@ +NVIDIA Tegra audio complex, with MAX98090 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-max98090" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the MAX98090's pins (as documented in its binding), and the jacks + on the board: + + * Headphones + * Speakers + * Mic Jack + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the MAX98090 audio codec. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in + +Example: + +sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 8fc653c..65a85f5 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -116,3 +116,13 @@ config SND_SOC_TEGRA_ALC5632 help Say Y or M here if you want to add support for SoC audio on the Toshiba AC100 netbook. + +config SND_SOC_TEGRA_MAX98090 + tristate "SoC Audio support for Tegra boards using a MAX98090 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_MAX98090 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the MAX98090 codec, such as Venice2. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 21d2550..5ae588c 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -24,6 +24,7 @@ snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o @@ -31,3 +32,4 @@ obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_MAX98090) += snd-soc-tegra-max98090.o diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c new file mode 100644 index 0000000..0283cfb --- /dev/null +++ b/sound/soc/tegra/tegra_max98090.c @@ -0,0 +1,275 @@ +/* + * Tegra machine ASoC driver for boards using a MAX90809 CODEC. + * + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-max98090" + +struct tegra_max98090 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; +}; + +static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + mclk = 12000000; + break; + } + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_max98090_ops = { + .hw_params = tegra_max98090_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_max98090_hp_jack; + +static struct snd_soc_jack_pin tegra_max98090_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, +}; + +static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack); + snd_soc_jack_add_pins(&tegra_max98090_hp_jack, + ARRAY_SIZE(tegra_max98090_hp_jack_pins), + tegra_max98090_hp_jack_pins); + + tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, + 1, + &tegra_max98090_hp_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_max98090_dai = { + .name = "max98090", + .stream_name = "max98090 PCM", + .codec_dai_name = "HiFi", + .init = tegra_max98090_asoc_init, + .ops = &tegra_max98090_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_max98090 = { + .name = "tegra-max98090", + .owner = THIS_MODULE, + .dai_link = &tegra_max98090_dai, + .num_links = 1, + .controls = tegra_max98090_controls, + .num_controls = ARRAY_SIZE(tegra_max98090_controls), + .dapm_widgets = tegra_max98090_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_max98090_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_max98090_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_max98090; + struct tegra_max98090 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_max98090), GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_max98090\n"); + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_max98090_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_max98090_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_max98090_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_max98090_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_max98090_dai.platform_of_node = tegra_max98090_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1, + &tegra_max98090_hp_jack_gpio); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_max98090_of_match[] = { + { .compatible = "nvidia,tegra-audio-max98090", }, + {}, +}; + +static struct platform_driver tegra_max98090_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_max98090_of_match, + }, + .probe = tegra_max98090_probe, + .remove = tegra_max98090_remove, +}; +module_platform_driver(tegra_max98090_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra max98090 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_max98090_of_match); -- cgit v0.10.2 From 308a0f3f24db5e5943ef0aad722e8b3d125dbddb Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 4 Dec 2013 15:19:26 -0700 Subject: ASoC: max98090: add DT binding document for MAX98090 CODEC This binding mainly serves to document the list of input and output pins that may be used in a sound card's audio routing table. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt new file mode 100644 index 0000000..e4c8b36 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -0,0 +1,43 @@ +MAX98090 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98090". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Pins on the device (for linking into audio routes): + + * MIC1 + * MIC2 + * DMICL + * DMICR + * IN1 + * IN2 + * IN3 + * IN4 + * IN5 + * IN6 + * IN12 + * IN34 + * IN56 + * HPL + * HPR + * SPKL + * SPKR + * RCVL + * RCVR + * MICBIAS + +Example: + +audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpio>; + interrupts = ; +}; -- cgit v0.10.2 From 00e6cb2aed48a86e97a244c6f96ac1f934e2272c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Dec 2013 11:02:49 +0100 Subject: dt: Add bindings documentation for the ADI AXI-I2S controller This patch adds the devicetree documentation for the ADI AXI-SPDIF audio controller. The controller has: * One set of memory mapped register * Two clocks, one for the memory mapped register interface, one used as the audio reference clock * One DMA interface each for the transmit and receive data Signed-off-by: Lars-Peter Clausen Cc: Rob Herring Cc: Pawel Moll Cc: Mark Rutland Cc: Stephen Warren Cc: Ian Campbell Cc: devicetree@vger.kernel.org Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 0000000..5875ca4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,31 @@ +ADI AXI-I2S controller + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + i2s: i2s@0x77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>, <&ps7_dma 1>; + dma-names = "tx", "rx"; + }; -- cgit v0.10.2 From 8f2fe346822419ee729c081b71c8835b733e0884 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Dec 2013 11:02:50 +0100 Subject: ASoC: Add support for the Analog Devices AXI-I2S core This patch adds support for the AXI-I2S softcore. The core implements a simple bidirectional I2S transceiver and is used by Analog Devices in some of their reference designs for various FPGA platforms. The driver uses the generic PCM dmaengine driver for its PCM. The only restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as the dmaengine driver for the DMA core (PL330) that is used with this core has no residue reporting capabilities yet. This will be fixed in the future though. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b84..866dfec 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM select SND_DMAENGINE_PCM # All the supported SoCs +source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e701..c70c7f7 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -8,6 +8,7 @@ endif obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ +obj-$(CONFIG_SND_SOC) += adi/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig new file mode 100644 index 0000000..46f4b79 --- /dev/null +++ b/sound/soc/adi/Kconfig @@ -0,0 +1,13 @@ +config SND_SOC_ADI + tristate "Audio support for Analog Devices reference designs" + depends on MICROBLAZE || ARCH_ZYNQ || COMPILE_TEST + help + Audio support for various reference designs by Analog Devices. + +config SND_SOC_ADI_AXI_I2S + tristate "AXI-I2S support" + depends on SND_SOC_ADI + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + ASoC driver for the Analog Devices AXI-I2S softcore peripheral. diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile new file mode 100644 index 0000000..d32c21a --- /dev/null +++ b/sound/soc/adi/Makefile @@ -0,0 +1,3 @@ +snd-soc-adi-axi-i2s-objs := axi-i2s.o + +obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c new file mode 100644 index 0000000..0822c77 --- /dev/null +++ b/sound/soc/adi/axi-i2s.c @@ -0,0 +1,277 @@ +/* + * Copyright (C) 2012-2013, Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#define AXI_I2S_REG_RESET 0x00 +#define AXI_I2S_REG_CTRL 0x04 +#define AXI_I2S_REG_CLK_CTRL 0x08 +#define AXI_I2S_REG_STATUS 0x10 + +#define AXI_I2S_REG_RX_FIFO 0x28 +#define AXI_I2S_REG_TX_FIFO 0x2C + +#define AXI_I2S_RESET_GLOBAL BIT(0) +#define AXI_I2S_RESET_TX_FIFO BIT(1) +#define AXI_I2S_RESET_RX_FIFO BIT(2) + +#define AXI_I2S_CTRL_TX_EN BIT(0) +#define AXI_I2S_CTRL_RX_EN BIT(1) + +/* The frame size is configurable, but for now we always set it 64 bit */ +#define AXI_I2S_BITS_PER_FRAME 64 + +struct axi_i2s { + struct regmap *regmap; + struct clk *clk; + struct clk *clk_ref; + + struct snd_soc_dai_driver dai_driver; + + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; + + struct snd_ratnum ratnum; + struct snd_pcm_hw_constraint_ratnums rate_constraints; +}; + +static int axi_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = AXI_I2S_CTRL_RX_EN; + else + mask = AXI_I2S_CTRL_TX_EN; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = mask; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, AXI_I2S_REG_CTRL, mask, val); + + return 0; +} + +static int axi_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int bclk_div, word_size; + unsigned int bclk_rate; + + bclk_rate = params_rate(params) * AXI_I2S_BITS_PER_FRAME; + + word_size = AXI_I2S_BITS_PER_FRAME / 2 - 1; + bclk_div = DIV_ROUND_UP(clk_get_rate(i2s->clk_ref), bclk_rate) / 2 - 1; + + regmap_write(i2s->regmap, AXI_I2S_REG_CLK_CTRL, (word_size << 16) | + bclk_div); + + return 0; +} + +static int axi_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + uint32_t mask; + int ret; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = AXI_I2S_RESET_RX_FIFO; + else + mask = AXI_I2S_RESET_TX_FIFO; + + regmap_write(i2s->regmap, AXI_I2S_REG_RESET, mask); + + ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &i2s->rate_constraints); + if (ret) + return ret; + + return clk_prepare_enable(i2s->clk_ref); +} + +static void axi_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(i2s->clk_ref); +} + +static int axi_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); + + return 0; +} + +static const struct snd_soc_dai_ops axi_i2s_dai_ops = { + .startup = axi_i2s_startup, + .shutdown = axi_i2s_shutdown, + .trigger = axi_i2s_trigger, + .hw_params = axi_i2s_hw_params, +}; + +static struct snd_soc_dai_driver axi_i2s_dai = { + .probe = axi_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, + }, + .ops = &axi_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static const struct snd_soc_component_driver axi_i2s_component = { + .name = "axi-i2s", +}; + +static const struct regmap_config axi_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AXI_I2S_REG_STATUS, +}; + +static int axi_i2s_probe(struct platform_device *pdev) +{ + struct resource *res; + struct axi_i2s *i2s; + void __iomem *base; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + platform_set_drvdata(pdev, i2s); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_request_and_ioremap(&pdev->dev, res); + if (!base) + return -EBUSY; + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &axi_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) + return PTR_ERR(i2s->regmap); + + i2s->clk = devm_clk_get(&pdev->dev, "axi"); + if (IS_ERR(i2s->clk)) + return PTR_ERR(i2s->clk); + + i2s->clk_ref = devm_clk_get(&pdev->dev, "ref"); + if (IS_ERR(i2s->clk_ref)) + return PTR_ERR(i2s->clk_ref); + + ret = clk_prepare_enable(i2s->clk); + if (ret) + return ret; + + i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO; + i2s->playback_dma_data.addr_width = 4; + i2s->playback_dma_data.maxburst = 1; + + i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO; + i2s->capture_dma_data.addr_width = 4; + i2s->capture_dma_data.maxburst = 1; + + i2s->ratnum.num = clk_get_rate(i2s->clk_ref) / 2 / AXI_I2S_BITS_PER_FRAME; + i2s->ratnum.den_step = 1; + i2s->ratnum.den_min = 1; + i2s->ratnum.den_max = 64; + + i2s->rate_constraints.rats = &i2s->ratnum; + i2s->rate_constraints.nrats = 1; + + regmap_write(i2s->regmap, AXI_I2S_REG_RESET, AXI_I2S_RESET_GLOBAL); + + ret = devm_snd_soc_register_component(&pdev->dev, &axi_i2s_component, + &axi_i2s_dai, 1); + if (ret) + goto err_clk_disable; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (ret) + goto err_clk_disable; + +err_clk_disable: + clk_disable_unprepare(i2s->clk); + return ret; +} + +static int axi_i2s_dev_remove(struct platform_device *pdev) +{ + struct axi_i2s *i2s = platform_get_drvdata(pdev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static const struct of_device_id axi_i2s_of_match[] = { + { .compatible = "adi,axi-i2s-1.00.a", }, + {}, +}; +MODULE_DEVICE_TABLE(of, axi_i2s_of_match); + +static struct platform_driver axi_i2s_driver = { + .driver = { + .name = "axi-i2s", + .owner = THIS_MODULE, + .of_match_table = axi_i2s_of_match, + }, + .probe = axi_i2s_probe, + .remove = axi_i2s_dev_remove, +}; +module_platform_driver(axi_i2s_driver); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("AXI I2S driver"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From d7b528eff9277b83b315500f44ade178035ed0d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Dec 2013 11:02:51 +0100 Subject: dt: Add bindings documentation for the ADI AXI-SPDIF audio controller This patch adds the devicetree documentation for the ADI AXI-SPDIF audio controller. The controller has: * One set of memory mapped register * Two clocks, one for the memory mapped register interface, one used as the audio reference clock * A DMA interface for the transmit data Signed-off-by: Lars-Peter Clausen Cc: Rob Herring Cc: Pawel Moll Cc: Mark Rutland Cc: Stephen Warren Cc: Ian Campbell Cc: devicetree@vger.kernel.org Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt new file mode 100644 index 0000000..46f3449 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt @@ -0,0 +1,30 @@ +ADI AXI-SPDIF controller + +Required properties: + - compatible : Must be "adi,axi-spdif-1.00.a" + - reg : Must contain SPDIF core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names: "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channel that is used by + the core. The core expects one dma channel for transmit. + - dma-names : Must be "tx" + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + spdif: spdif@0x77400000 { + compatible = "adi,axi-spdif-tx-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>; + dma-names = "tx"; + }; -- cgit v0.10.2 From 429e4374cc007f260c7d0e2d0df5247deeaf8fbe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Dec 2013 11:02:52 +0100 Subject: ASoC: Add support for the Analog Devices AXI-SPDIF driver This patch adds a ASoC driver for the AXI-SPDIF softcore. The core implements a simple SPDIF transmitter and is used on some Analog Devices' reference designs for various FPGA platforms. For now the driver only support the PL330 as the the DMA controller. The driver uses the generic PCM dmaengine driver for its PCM. The only restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as the dmaengine driver for the DMA core (PL330) that is used with this core has no residue reporting capabilities yet. This will be fixed in the future though. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig index 46f4b79..dd763f5 100644 --- a/sound/soc/adi/Kconfig +++ b/sound/soc/adi/Kconfig @@ -11,3 +11,11 @@ config SND_SOC_ADI_AXI_I2S select REGMAP_MMIO help ASoC driver for the Analog Devices AXI-I2S softcore peripheral. + +config SND_SOC_ADI_AXI_SPDIF + tristate "AXI-SPDIF support" + depends on SND_SOC_ADI + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + ASoC driver for the Analog Devices AXI-SPDIF softcore peripheral. diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile index d32c21a..64456c1 100644 --- a/sound/soc/adi/Makefile +++ b/sound/soc/adi/Makefile @@ -1,3 +1,5 @@ snd-soc-adi-axi-i2s-objs := axi-i2s.o +snd-soc-adi-axi-spdif-objs := axi-spdif.o obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o +obj-$(CONFIG_SND_SOC_ADI_AXI_SPDIF) += snd-soc-adi-axi-spdif.o diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c new file mode 100644 index 0000000..d5408d2 --- /dev/null +++ b/sound/soc/adi/axi-spdif.c @@ -0,0 +1,272 @@ +/* + * Copyright (C) 2012-2013, Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#define AXI_SPDIF_REG_CTRL 0x0 +#define AXI_SPDIF_REG_STAT 0x4 +#define AXI_SPDIF_REG_TX_FIFO 0xc + +#define AXI_SPDIF_CTRL_TXDATA BIT(1) +#define AXI_SPDIF_CTRL_TXEN BIT(0) +#define AXI_SPDIF_CTRL_CLKDIV_OFFSET 8 +#define AXI_SPDIF_CTRL_CLKDIV_MASK (0xff << 8) + +#define AXI_SPDIF_FREQ_44100 (0x0 << 6) +#define AXI_SPDIF_FREQ_48000 (0x1 << 6) +#define AXI_SPDIF_FREQ_32000 (0x2 << 6) +#define AXI_SPDIF_FREQ_NA (0x3 << 6) + +struct axi_spdif { + struct regmap *regmap; + struct clk *clk; + struct clk *clk_ref; + + struct snd_dmaengine_dai_dma_data dma_data; + + struct snd_ratnum ratnum; + struct snd_pcm_hw_constraint_ratnums rate_constraints; +}; + +static int axi_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = AXI_SPDIF_CTRL_TXDATA; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXDATA, val); + + return 0; +} + +static int axi_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int clkdiv, stat; + + switch (params_rate(params)) { + case 32000: + stat = AXI_SPDIF_FREQ_32000; + break; + case 44100: + stat = AXI_SPDIF_FREQ_44100; + break; + case 48000: + stat = AXI_SPDIF_FREQ_48000; + break; + default: + stat = AXI_SPDIF_FREQ_NA; + break; + } + + clkdiv = DIV_ROUND_CLOSEST(clk_get_rate(spdif->clk_ref), + rate * 64 * 2) - 1; + clkdiv <<= AXI_SPDIF_CTRL_CLKDIV_OFFSET; + + regmap_write(spdif->regmap, AXI_SPDIF_REG_STAT, stat); + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_CLKDIV_MASK, clkdiv); + + return 0; +} + +static int axi_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &spdif->dma_data, NULL); + + return 0; +} + +static int axi_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &spdif->rate_constraints); + if (ret) + return ret; + + ret = clk_prepare_enable(spdif->clk_ref); + if (ret) + return ret; + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN); + + return 0; +} + +static void axi_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXEN, 0); + + clk_disable_unprepare(spdif->clk_ref); +} + +static const struct snd_soc_dai_ops axi_spdif_dai_ops = { + .startup = axi_spdif_startup, + .shutdown = axi_spdif_shutdown, + .trigger = axi_spdif_trigger, + .hw_params = axi_spdif_hw_params, +}; + +static struct snd_soc_dai_driver axi_spdif_dai = { + .probe = axi_spdif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &axi_spdif_dai_ops, +}; + +static const struct snd_soc_component_driver axi_spdif_component = { + .name = "axi-spdif", +}; + +static const struct regmap_config axi_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AXI_SPDIF_REG_STAT, +}; + +static int axi_spdif_probe(struct platform_device *pdev) +{ + struct axi_spdif *spdif; + struct resource *res; + void __iomem *base; + int ret; + + spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL); + if (!spdif) + return -ENOMEM; + + platform_set_drvdata(pdev, spdif); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_request_and_ioremap(&pdev->dev, res); + if (!base) + return -EBUSY; + + spdif->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &axi_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) + return PTR_ERR(spdif->regmap); + + spdif->clk = devm_clk_get(&pdev->dev, "axi"); + if (IS_ERR(spdif->clk)) + return PTR_ERR(spdif->clk); + + spdif->clk_ref = devm_clk_get(&pdev->dev, "ref"); + if (IS_ERR(spdif->clk_ref)) + return PTR_ERR(spdif->clk_ref); + + ret = clk_prepare_enable(spdif->clk); + if (ret) + return ret; + + spdif->dma_data.addr = res->start + AXI_SPDIF_REG_TX_FIFO; + spdif->dma_data.addr_width = 4; + spdif->dma_data.maxburst = 1; + + spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128; + spdif->ratnum.den_step = 1; + spdif->ratnum.den_min = 1; + spdif->ratnum.den_max = 64; + + spdif->rate_constraints.rats = &spdif->ratnum; + spdif->rate_constraints.nrats = 1; + + ret = devm_snd_soc_register_component(&pdev->dev, &axi_spdif_component, + &axi_spdif_dai, 1); + if (ret) + goto err_clk_disable; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (ret) + goto err_clk_disable; + + return 0; + +err_clk_disable: + clk_disable_unprepare(spdif->clk); + return ret; +} + +static int axi_spdif_dev_remove(struct platform_device *pdev) +{ + struct axi_spdif *spdif = platform_get_drvdata(pdev); + + clk_disable_unprepare(spdif->clk); + + return 0; +} + +static const struct of_device_id axi_spdif_of_match[] = { + { .compatible = "adi,axi-spdif-tx-1.00.a", }, + {}, +}; +MODULE_DEVICE_TABLE(of, axi_spdif_of_match); + +static struct platform_driver axi_spdif_driver = { + .driver = { + .name = "axi-spdif", + .owner = THIS_MODULE, + .of_match_table = axi_spdif_of_match, + }, + .probe = axi_spdif_probe, + .remove = axi_spdif_dev_remove, +}; +module_platform_driver(axi_spdif_driver); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("AXI SPDIF driver"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From 58381da687742a3d8bbb98362152de8326a0c077 Mon Sep 17 00:00:00 2001 From: Jan Weitzel Date: Thu, 5 Dec 2013 09:54:02 +0100 Subject: ASoC: tlv320aic3x: no mono controls 3007 model if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite registers for class-d amplifier. classd amplifier controls are only used for AIC3X_MODEL_3007. Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007 codecs. Testet only with AIC3X_MODEL_3007 Signed-off-by: Jan Weitzel Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 546d16b..470fbfb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -350,16 +350,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", - LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", - PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", - DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -383,7 +373,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output pin mute controls */ SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), - SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, @@ -412,6 +401,20 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; +static const struct snd_kcontrol_new aic3x_mono_controls[] = { + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), +}; + /* * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps */ @@ -565,9 +568,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), - /* Mono Output */ - SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, @@ -626,9 +626,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_line_mixer_controls[0], ARRAY_SIZE(aic3x_right_line_mixer_controls)), - SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_mono_mixer_controls[0], - ARRAY_SIZE(aic3x_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_hp_mixer_controls[0], ARRAY_SIZE(aic3x_left_hp_mixer_controls)), @@ -644,7 +641,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LLOUT"), SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("MONO_LOUT"), SND_SOC_DAPM_OUTPUT("HPLOUT"), SND_SOC_DAPM_OUTPUT("HPROUT"), SND_SOC_DAPM_OUTPUT("HPLCOM"), @@ -666,6 +662,17 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Detection"), }; +static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_mono_mixer_controls[0], + ARRAY_SIZE(aic3x_mono_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), +}; + static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { /* Class-D outputs */ SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), @@ -754,17 +761,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right DAC Mux"}, {"RLOUT", NULL, "Right Line Out"}, - /* Mono Output */ - {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, - {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, - {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, - {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, - {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, - - {"Mono Out", NULL, "Mono Mixer"}, - {"MONO_LOUT", NULL, "Mono Out"}, - /* Left HP Output */ {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, @@ -820,6 +816,18 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +static const struct snd_soc_dapm_route intercon_mono[] = { + /* Mono Output */ + {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, + {"Mono Out", NULL, "Mono Mixer"}, + {"MONO_LOUT", NULL, "Mono Out"}, +}; + static const struct snd_soc_dapm_route intercon_3007[] = { /* Class-D outputs */ {"Left Class-D Out", NULL, "Left Line Out"}, @@ -833,11 +841,20 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - if (aic3x->model == AIC3X_MODEL_3007) { + switch (aic3x->model) { + case AIC3X_MODEL_3X: + case AIC3X_MODEL_33: + snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, + ARRAY_SIZE(aic3x_dapm_mono_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_mono, + ARRAY_SIZE(intercon_mono)); + break; + case AIC3X_MODEL_3007: snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); + break; } return 0; @@ -1218,6 +1235,24 @@ static int aic3x_resume(struct snd_soc_codec *codec) return 0; } +static void aic3x_mono_init(struct snd_soc_codec *codec) +{ + /* DAC to Mono Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute all outputs */ + snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); + + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); +} + /* * initialise the AIC3X driver * register the mixer and dsp interfaces with the kernel @@ -1241,14 +1276,10 @@ static int aic3x_init(struct snd_soc_codec *codec) /* DAC to Line Out default volume and route to Output mixer */ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE); - snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE); @@ -1269,9 +1300,6 @@ static int aic3x_init(struct snd_soc_codec *codec) /* PGA to Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); @@ -1281,12 +1309,15 @@ static int aic3x_init(struct snd_soc_codec *codec) /* Line2 Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); - if (aic3x->model == AIC3X_MODEL_3007) { + switch (aic3x->model) { + case AIC3X_MODEL_3X: + case AIC3X_MODEL_33: + aic3x_mono_init(codec); + break; + case AIC3X_MODEL_3007: snd_soc_write(codec, CLASSD_CTRL, 0); + break; } return 0; @@ -1343,8 +1374,17 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - if (aic3x->model == AIC3X_MODEL_3007) - snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + switch (aic3x->model) { + case AIC3X_MODEL_3X: + case AIC3X_MODEL_33: + snd_soc_add_codec_controls(codec, aic3x_mono_controls, + ARRAY_SIZE(aic3x_mono_controls)); + break; + case AIC3X_MODEL_3007: + snd_soc_add_codec_controls(codec, + &aic3x_classd_amp_gain_ctrl, 1); + break; + } /* set mic bias voltage */ switch (aic3x->micbias_vg) { -- cgit v0.10.2 From 1a1c75a7982ca1181bb84bb9e83f9f7c752fb104 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Dec 2013 17:26:01 -0800 Subject: ASoC: rsnd: gen: fixup Gen2 channel size Gen2 has 0 - 9, total 10 channels, not 9 channels. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index bf066f7..d0ab203 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -217,11 +217,11 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, /* single address mapping */ #define RSND_GEN2_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9) + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10) /* multi address mapping */ #define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9) + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10) static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { -- cgit v0.10.2 From 860d0c0dd2e7dbf98b47f38d80793137bc6c6ebc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:17 +0200 Subject: ASoC: davinci: Kconfig: Remove help section for SND_DAVINCI_SOC The help text is misleading and the prompt itself explains the purpose of this config section. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 95970f5..fb91826 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,11 +1,6 @@ config SND_DAVINCI_SOC tristate "SoC Audio for the TI DAVINCI or AM33XX chip" depends on ARCH_DAVINCI || SOC_AM33XX - help - Platform driver for daVinci or AM33xx - Say Y or M if you want to add support for codecs attached to - the DAVINCI AC97, I2S, or McASP interface. You will also need - to select the audio interfaces to support below. config SND_DAVINCI_SOC_I2S tristate -- cgit v0.10.2 From c3238a4c058edd1528f0bec9a37fe79479e9e1a8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:18 +0200 Subject: ASoC: davinci: Kconfig/Makefile: Generic EVM machine driver related cleanup We have several boards using the same machine driver for audio support. All of these machines can select a generic machine driver config option to build the needed driver while keeping the config options used within the driver for compile time code path selection. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index fb91826..be66771 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -11,11 +11,15 @@ config SND_DAVINCI_SOC_MCASP config SND_DAVINCI_SOC_VCIF tristate +config SND_DAVINCI_SOC_GENERIC_EVM + tristate + select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_MCASP + config SND_AM33XX_SOC_EVM tristate "SoC Audio for the AM33XX chip based boards" depends on SND_DAVINCI_SOC && SOC_AM33XX - select SND_SOC_TLV320AIC3X - select SND_DAVINCI_SOC_MCASP + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y or M if you want to add support for SoC audio on AM33XX boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, @@ -26,8 +30,7 @@ config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM - select SND_DAVINCI_SOC_I2S - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI DaVinci DM6446, DM355 or DM365 EVM platforms. @@ -54,8 +57,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM - select SND_DAVINCI_SOC_MCASP - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF help @@ -64,8 +66,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM - select SND_DAVINCI_SOC_MCASP - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI @@ -74,8 +75,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM - select SND_DAVINCI_SOC_MCASP - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI DA850/OMAP-L138 EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index bc81e79..744d4d9 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -9,11 +9,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o -# DAVINCI Machine Support +# Generic DAVINCI/AM33xx Machine Support snd-soc-evm-objs := davinci-evm.o -obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_AM33XX_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_GENERIC_EVM) += snd-soc-evm.o -- cgit v0.10.2 From a42efd97f7471c78617c6329ed39919e2f31a7cc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:19 +0200 Subject: ASoC: davinci: kconfig: Prepare for AM43xx support AM43xx have the same McASP IP as AM33xx and both platform uses eDMA. Modify the Kconfig so it will be possible to add audio support for AM43xx based boards later. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index ed785b3..1eed972 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,7 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, TI81xx) + "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - reg : Should contain reg specifiers for the entries in the reg-names property. - reg-names : Should contain: diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index be66771..a8ec1fc 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,6 +1,6 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for the TI DAVINCI or AM33XX chip" - depends on ARCH_DAVINCI || SOC_AM33XX + tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips" + depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX config SND_DAVINCI_SOC_I2S tristate -- cgit v0.10.2 From 7b6a772932eca1ed8d22414a1ba985c5232369d0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:20 +0200 Subject: ASoC: davinci-evm: Do not include davinci-mcasp.h There's no need to include this header file here. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 623eb5e..2a00e2d 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,7 +28,6 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" -#include "davinci-mcasp.h" struct snd_soc_card_drvdata_davinci { unsigned sysclk; -- cgit v0.10.2 From d38970e1363ddf63fa2f681f0c9b47fc91ca3961 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:21 +0200 Subject: ASoC: davinci-evm: Switch to use .dai_fmt of snd_soc_dai_link(s) Specify the dai formats to use within the snd_soc_dai_link structures. In this way we can remove the code dealing with the dai format configuration from the machin driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 2a00e2d..70ff377 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,6 @@ struct snd_soc_card_drvdata_davinci { unsigned sysclk; }; -#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ - SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -47,16 +45,6 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) snd_soc_card_get_drvdata(soc_card))->sysclk; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - /* set the codec system clock */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) @@ -70,24 +58,10 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int evm_spdif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - /* set cpu DAI configuration */ - return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); -} - static struct snd_soc_ops evm_ops = { .hw_params = evm_hw_params, }; -static struct snd_soc_ops evm_spdif_ops = { - .hw_params = evm_spdif_hw_params, -}; - /* davinci-evm machine dapm widgets */ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -164,6 +138,8 @@ static struct snd_soc_dai_link dm6446_evm_dai = { .platform_name = "davinci-mcbsp", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct snd_soc_dai_link dm355_evm_dai = { @@ -175,6 +151,8 @@ static struct snd_soc_dai_link dm355_evm_dai = { .platform_name = "davinci-mcbsp.1", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct snd_soc_dai_link dm365_evm_dai = { @@ -183,10 +161,12 @@ static struct snd_soc_dai_link dm365_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", - .init = evm_aic3x_init, .codec_name = "tlv320aic3x-codec.1-0018", - .ops = &evm_ops, .platform_name = "davinci-mcbsp", + .init = evm_aic3x_init, + .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", @@ -207,6 +187,8 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .codec_name = "tlv320aic3x-codec.0-001a", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }, { .name = "McASP", @@ -215,7 +197,8 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .codec_dai_name = "dit-hifi", .codec_name = "spdif_dit", .platform_name = "davinci-mcasp.1", - .ops = &evm_spdif_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }, }; @@ -228,6 +211,8 @@ static struct snd_soc_dai_link da830_evm_dai = { .platform_name = "davinci-mcasp.1", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct snd_soc_dai_link da850_evm_dai = { @@ -239,6 +224,8 @@ static struct snd_soc_dai_link da850_evm_dai = { .platform_name = "davinci-mcasp.0", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; /* davinci dm6446 evm audio machine driver */ @@ -335,6 +322,8 @@ static struct snd_soc_dai_link evm_dai_tlv320aic3x = { .codec_dai_name = "tlv320aic3x-hifi", .ops = &evm_ops, .init = evm_aic3x_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static const struct of_device_id davinci_evm_dt_ids[] = { -- cgit v0.10.2 From ed29cd5e8d9941353f82784a2478ba6babc828da Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:22 +0200 Subject: ASoC: davinci-mcasp: Move DAVINCI_MCASP_RATE from header to source file It is not used outside of the .c file. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 71e14bb3..9763a5d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -955,6 +955,8 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_sysclk = davinci_mcasp_set_sysclk, }; +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 + #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a2e27e1..a84e796 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -23,7 +23,6 @@ #include "davinci-pcm.h" -#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_I2S_DAI 0 #define DAVINCI_MCASP_DIT_DAI 1 -- cgit v0.10.2 From 18f93506623aacbb269f47cbda9fe90ffc5acda6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:23 +0200 Subject: ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_I2S/DIT_DAI defines These are not used, probably leftovers from the past. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a84e796..70b089b 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -23,9 +23,6 @@ #include "davinci-pcm.h" -#define DAVINCI_MCASP_I2S_DAI 0 -#define DAVINCI_MCASP_DIT_DAI 1 - struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; -- cgit v0.10.2 From 58e48d9774d4d8fc5e0785dbd2ccf075b248ad96 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:24 +0200 Subject: ASoC: davinci-mcasp: Correct dai driver struct initialization for 2nd dai Add .name when assigning the dai name. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9763a5d..6cde1ba 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -987,7 +987,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }, { - "davinci-mcasp.1", + .name = "davinci-mcasp.1", .playback = { .channels_min = 1, .channels_max = 384, -- cgit v0.10.2 From 02e08d9b6bd67784d4c58e659c21674b31972c34 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:25 +0200 Subject: ASoC: davinci-mcasp: Move register definitions to header file It is better for readability to have the register definitions out from the source file. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6cde1ba..1c1585e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -35,273 +35,6 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" -/* - * McASP register definitions - */ -#define DAVINCI_MCASP_PID_REG 0x00 -#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 - -#define DAVINCI_MCASP_PFUNC_REG 0x10 -#define DAVINCI_MCASP_PDIR_REG 0x14 -#define DAVINCI_MCASP_PDOUT_REG 0x18 -#define DAVINCI_MCASP_PDSET_REG 0x1c - -#define DAVINCI_MCASP_PDCLR_REG 0x20 - -#define DAVINCI_MCASP_TLGC_REG 0x30 -#define DAVINCI_MCASP_TLMR_REG 0x34 - -#define DAVINCI_MCASP_GBLCTL_REG 0x44 -#define DAVINCI_MCASP_AMUTE_REG 0x48 -#define DAVINCI_MCASP_LBCTL_REG 0x4c - -#define DAVINCI_MCASP_TXDITCTL_REG 0x50 - -#define DAVINCI_MCASP_GBLCTLR_REG 0x60 -#define DAVINCI_MCASP_RXMASK_REG 0x64 -#define DAVINCI_MCASP_RXFMT_REG 0x68 -#define DAVINCI_MCASP_RXFMCTL_REG 0x6c - -#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 -#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 -#define DAVINCI_MCASP_RXTDM_REG 0x78 -#define DAVINCI_MCASP_EVTCTLR_REG 0x7c - -#define DAVINCI_MCASP_RXSTAT_REG 0x80 -#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 -#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 -#define DAVINCI_MCASP_REVTCTL_REG 0x8c - -#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 -#define DAVINCI_MCASP_TXMASK_REG 0xa4 -#define DAVINCI_MCASP_TXFMT_REG 0xa8 -#define DAVINCI_MCASP_TXFMCTL_REG 0xac - -#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 -#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 -#define DAVINCI_MCASP_TXTDM_REG 0xb8 -#define DAVINCI_MCASP_EVTCTLX_REG 0xbc - -#define DAVINCI_MCASP_TXSTAT_REG 0xc0 -#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 -#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 -#define DAVINCI_MCASP_XEVTCTL_REG 0xcc - -/* Left(even TDM Slot) Channel Status Register File */ -#define DAVINCI_MCASP_DITCSRA_REG 0x100 -/* Right(odd TDM slot) Channel Status Register File */ -#define DAVINCI_MCASP_DITCSRB_REG 0x118 -/* Left(even TDM slot) User Data Register File */ -#define DAVINCI_MCASP_DITUDRA_REG 0x130 -/* Right(odd TDM Slot) User Data Register File */ -#define DAVINCI_MCASP_DITUDRB_REG 0x148 - -/* Serializer n Control Register */ -#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 -#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ - (n << 2)) - -/* Transmit Buffer for Serializer n */ -#define DAVINCI_MCASP_TXBUF_REG 0x200 -/* Receive Buffer for Serializer n */ -#define DAVINCI_MCASP_RXBUF_REG 0x280 - -/* McASP FIFO Registers */ -#define DAVINCI_MCASP_WFIFOCTL (0x1010) -#define DAVINCI_MCASP_WFIFOSTS (0x1014) -#define DAVINCI_MCASP_RFIFOCTL (0x1018) -#define DAVINCI_MCASP_RFIFOSTS (0x101C) -#define MCASP_VER3_WFIFOCTL (0x1000) -#define MCASP_VER3_WFIFOSTS (0x1004) -#define MCASP_VER3_RFIFOCTL (0x1008) -#define MCASP_VER3_RFIFOSTS (0x100C) - -/* - * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management - * Register Bits - */ -#define MCASP_FREE BIT(0) -#define MCASP_SOFT BIT(1) - -/* - * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits - */ -#define AXR(n) (1< Date: Thu, 14 Nov 2013 11:35:26 +0200 Subject: ASoC: davinci-mcasp: Move private struct definition to source file Since it is a private struct strictly used by the davinci-mcasp driver it can be moved from header file to the source file. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1c1585e..7010795 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -37,6 +37,36 @@ #define DAVINCI_MCASP_NUM_SERIALIZER 16 +struct davinci_audio_dev { + struct davinci_pcm_dma_params dma_params[2]; + void __iomem *base; + struct device *dev; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; + u8 version; + u16 bclk_lrclk_ratio; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif +}; + static inline void mcasp_set_bits(void __iomem *reg, u32 val) { __raw_writel(__raw_readl(reg) | val, reg); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 619b98b..80e5a18 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -18,11 +18,6 @@ #ifndef DAVINCI_MCASP_H #define DAVINCI_MCASP_H -#include -#include - -#include "davinci-pcm.h" - /* * McASP register definitions */ @@ -290,34 +285,4 @@ #define NUMEVT_MASK (0xFF << 8) #define NUMDMA_MASK (0xFF) -struct davinci_audio_dev { - struct davinci_pcm_dma_params dma_params[2]; - void __iomem *base; - struct device *dev; - - /* McASP specific data */ - int tdm_slots; - u8 op_mode; - u8 num_serializer; - u8 *serial_dir; - u8 version; - u16 bclk_lrclk_ratio; - - /* McASP FIFO related */ - u8 txnumevt; - u8 rxnumevt; - -#ifdef CONFIG_PM_SLEEP - struct { - u32 txfmtctl; - u32 rxfmtctl; - u32 txfmt; - u32 rxfmt; - u32 aclkxctl; - u32 aclkrctl; - u32 pdir; - } context; -#endif -}; - #endif /* DAVINCI_MCASP_H */ -- cgit v0.10.2 From 57f439b8676b2dad63a148de4e61d80b4e196c2a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:27 +0200 Subject: ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_NUM_SERIALIZER define It is not used in the code. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7010795..e4c0fb4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -35,8 +35,6 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" -#define DAVINCI_MCASP_NUM_SERIALIZER 16 - struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; -- cgit v0.10.2 From eba0ecf067913d60768bb3d11f861b949f072a93 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:28 +0200 Subject: ASoC: davinci-mcasp: Do not inline the mcasp_set_ctl_reg function It brings no benefit to inline this function due to it's size and function. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e4c0fb4..ce1607b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -90,7 +90,7 @@ static inline u32 mcasp_get_reg(void __iomem *reg) return (unsigned int)__raw_readl(reg); } -static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) { int i = 0; -- cgit v0.10.2 From 70091a3e6aa2e7a05eaefcaec1a43c27a5023eb7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:29 +0200 Subject: ASoC: davinci-mcasp: Rename private struct and it's users (dev -> mcasp) Rename the private struct from davinci_audio_dev to davinci_mcasp. Change the local use of the pointer to this struct from *dev to *mcasp. The aim is to have better readable code for the first look since having dev->xxxx in the code when using the local private struct is a bit surprising. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ce1607b..bd85c98 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -35,7 +35,7 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" -struct davinci_audio_dev { +struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; struct device *dev; @@ -107,36 +107,36 @@ static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) printk(KERN_ERR "GBLCTL write error\n"); } -static void mcasp_start_rx(struct davinci_audio_dev *dev) +static void mcasp_start_rx(struct davinci_mcasp *mcasp) { - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); } -static void mcasp_start_tx(struct davinci_audio_dev *dev) +static void mcasp_start_tx(struct davinci_mcasp *mcasp) { u8 offset = 0, i; u32 cnt; - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); - for (i = 0; i < dev->num_serializer; i++) { - if (dev->serial_dir[i] == TX_MODE) { + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < mcasp->num_serializer; i++) { + if (mcasp->serial_dir[i] == TX_MODE) { offset = i; break; } @@ -144,116 +144,116 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) /* wait for TX ready */ cnt = 0; - while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + while (!(mcasp_get_reg(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & TXSTATE) && (cnt < 100000)) cnt++; - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); } -static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) +static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) { /* enable FIFO */ - switch (dev->version) { + if (mcasp->txnumevt) { /* enable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + FIFO_ENABLE); + mcasp_set_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); } } - mcasp_start_tx(dev); + mcasp_start_tx(mcasp); } else { - if (dev->rxnumevt) { /* enable FIFO */ - switch (dev->version) { + if (mcasp->rxnumevt) { /* enable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + FIFO_ENABLE); + mcasp_set_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } - mcasp_start_rx(dev); + mcasp_start_rx(mcasp); } } -static void mcasp_stop_rx(struct davinci_audio_dev *dev) +static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { - mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); } -static void mcasp_stop_tx(struct davinci_audio_dev *dev) +static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { - mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); } -static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) +static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) { /* disable FIFO */ - switch (dev->version) { + if (mcasp->txnumevt) { /* disable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); } } - mcasp_stop_tx(dev); + mcasp_stop_tx(mcasp); } else { - if (dev->rxnumevt) { /* disable FIFO */ - switch (dev->version) { + if (mcasp->rxnumevt) { /* disable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } - mcasp_stop_rx(dev); + mcasp_stop_rx(mcasp); } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - void __iomem *base = dev->base; + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + void __iomem *base = mcasp->base; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: - mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); break; default: /* configure a full-word SYNC pulse (LRCLK) */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); /* make 1st data bit occur one ACLK cycle after the frame sync */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); break; } @@ -342,25 +342,25 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); switch (div_id) { case 0: /* MCLK divider */ - mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; case 1: /* BCLK divider */ - mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKRCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); break; case 2: /* BCLK/LRCLK ratio */ - dev->bclk_lrclk_ratio = div; + mcasp->bclk_lrclk_ratio = div; break; default: @@ -373,22 +373,22 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (dir == SND_SOC_CLOCK_OUT) { - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { - mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); } return 0; } -static int davinci_config_channel_size(struct davinci_audio_dev *dev, +static int davinci_config_channel_size(struct davinci_mcasp *mcasp, int word_length) { u32 fmt; @@ -405,70 +405,70 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, * both left and right channels), so it has to be divided by number of * tdm-slots (for I2S - divided by 2). */ - if (dev->bclk_lrclk_ratio) - word_length = dev->bclk_lrclk_ratio / dev->tdm_slots; + if (mcasp->bclk_lrclk_ratio) + word_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), RXROT(7)); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXMASK_REG, mask); } - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXMASK_REG, mask); return 0; } -static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, +static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; u8 ser; - u8 slots = dev->tdm_slots; + u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); /* All PINS as McASP */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); } else { - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS); } - for (i = 0; i < dev->num_serializer; i++) { - mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), - dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE && + for (i = 0; i < mcasp->num_serializer; i++) { + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp->serial_dir[i]); + if (mcasp->serial_dir[i] == TX_MODE && tx_ser < max_active_serializers) { - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE && + } else if (mcasp->serial_dir[i] == RX_MODE && rx_ser < max_active_serializers) { - mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; } else { - mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), SRMOD_INACTIVE, SRMOD_MASK); } } @@ -479,127 +479,127 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, ser = rx_ser; if (ser < max_active_serializers) { - dev_warn(dev->dev, "stream has more channels (%d) than are " + dev_warn(mcasp->dev, "stream has more channels (%d) than are " "enabled in mcasp (%d)\n", channels, ser * slots); return -EINVAL; } - if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt * tx_ser > 64) - dev->txnumevt = 1; + if (mcasp->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (mcasp->txnumevt * tx_ser > 64) + mcasp->txnumevt = 1; - switch (dev->version) { + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, + mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, - ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); break; default: - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, - ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, + ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); } } - if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { - if (dev->rxnumevt * rx_ser > 64) - dev->rxnumevt = 1; - switch (dev->version) { + if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (mcasp->rxnumevt * rx_ser > 64) + mcasp->rxnumevt = 1; + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, + mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, - ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); break; default: - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, - ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, + ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } return 0; } -static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) +static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; - active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXORD); - if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); else printk(KERN_ERR "playback tdm slot %d not supported\n", - dev->tdm_slots); + mcasp->tdm_slots); } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask); - if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); else printk(KERN_ERR "capture tdm slot %d not supported\n", - dev->tdm_slots); + mcasp->tdm_slots); } } /* S/PDIF */ -static void davinci_hw_dit_param(struct davinci_audio_dev *dev) +static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); /* Set the TX tdm : for all the slots */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); /* Set the TX clock controls : div = 1 and internal */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE | TX_ASYNC); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); /* Enable the DIT */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; + &mcasp->dma_params[substream->stream]; int word_length; u8 fifo_level; - u8 slots = dev->tdm_slots; + u8 slots = mcasp->tdm_slots; u8 active_serializers; int channels; struct snd_interval *pcm_channels = hw_param_interval(params, @@ -608,17 +608,17 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, active_serializers = (channels + slots - 1) / slots; - if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt * active_serializers; + fifo_level = mcasp->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt * active_serializers; + fifo_level = mcasp->rxnumevt * active_serializers; - if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) - davinci_hw_dit_param(dev); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(mcasp); else - davinci_hw_param(dev, substream->stream); + davinci_hw_param(mcasp, substream->stream); switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: @@ -652,13 +652,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2 && !fifo_level) + if (mcasp->version == MCASP_VERSION_2 && !fifo_level) dma_params->acnt = 4; else dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; - davinci_config_channel_size(dev, word_length); + davinci_config_channel_size(mcasp, word_length); return 0; } @@ -666,29 +666,29 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = pm_runtime_get_sync(dev->dev); + ret = pm_runtime_get_sync(mcasp->dev); if (IS_ERR_VALUE(ret)) - dev_err(dev->dev, "pm_runtime_get_sync() failed\n"); - davinci_mcasp_start(dev, substream->stream); + dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n"); + davinci_mcasp_start(mcasp, substream->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: - davinci_mcasp_stop(dev, substream->stream); - ret = pm_runtime_put_sync(dev->dev); + davinci_mcasp_stop(mcasp, substream->stream); + ret = pm_runtime_put_sync(mcasp->dev); if (IS_ERR_VALUE(ret)) - dev_err(dev->dev, "pm_runtime_put_sync() failed\n"); + dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n"); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_mcasp_stop(dev, substream->stream); + davinci_mcasp_stop(mcasp, substream->stream); break; default: @@ -701,9 +701,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params); return 0; } @@ -915,7 +915,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct davinci_pcm_dma_params *dma_data; struct resource *mem, *ioarea, *res, *dat; struct snd_platform_data *pdata; - struct davinci_audio_dev *dev; + struct davinci_mcasp *mcasp; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -923,9 +923,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return -EINVAL; } - dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev), + mcasp = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcasp), GFP_KERNEL); - if (!dev) + if (!mcasp) return -ENOMEM; pdata = davinci_mcasp_set_pdata_from_of(pdev); @@ -936,7 +936,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!mem) { - dev_warn(dev->dev, + dev_warn(mcasp->dev, "\"mpu\" mem resource not found, using index 0\n"); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -960,27 +960,27 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return ret; } - dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!dev->base) { + mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!mcasp->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; goto err_release_clk; } - dev->op_mode = pdata->op_mode; - dev->tdm_slots = pdata->tdm_slots; - dev->num_serializer = pdata->num_serializer; - dev->serial_dir = pdata->serial_dir; - dev->version = pdata->version; - dev->txnumevt = pdata->txnumevt; - dev->rxnumevt = pdata->rxnumevt; - dev->dev = &pdev->dev; + mcasp->op_mode = pdata->op_mode; + mcasp->tdm_slots = pdata->tdm_slots; + mcasp->num_serializer = pdata->num_serializer; + mcasp->serial_dir = pdata->serial_dir; + mcasp->version = pdata->version; + mcasp->txnumevt = pdata->txnumevt; + mcasp->rxnumevt = pdata->rxnumevt; + mcasp->dev = &pdev->dev; dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (!dat) dat = mem; - dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; @@ -993,7 +993,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->channel = pdata->tx_dma_channel; - dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; @@ -1006,7 +1006,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->channel = pdata->rx_dma_channel; - dev_set_drvdata(&pdev->dev, dev); + dev_set_drvdata(&pdev->dev, mcasp); ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); @@ -1044,32 +1044,32 @@ static int davinci_mcasp_remove(struct platform_device *pdev) #ifdef CONFIG_PM_SLEEP static int davinci_mcasp_suspend(struct device *dev) { - struct davinci_audio_dev *a = dev_get_drvdata(dev); - void __iomem *base = a->base; + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + void __iomem *base = mcasp->base; - a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); - a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); - a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); - a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); - a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); - a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); - a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + mcasp->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + mcasp->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + mcasp->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + mcasp->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + mcasp->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + mcasp->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + mcasp->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); return 0; } static int davinci_mcasp_resume(struct device *dev) { - struct davinci_audio_dev *a = dev_get_drvdata(dev); - void __iomem *base = a->base; - - mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); - mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + void __iomem *base = mcasp->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); return 0; } -- cgit v0.10.2 From 8f113b77b511c9e26706d4eb077af0ba30893ee4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:30 +0200 Subject: ASoC: davinci-mcasp: Be consistent with the use of base in davinci_mcasp_set_dai_fmt Replace mcasp->base use with plain base in the davinci_mcasp_set_dai_fmt() function since it has been already used by the remaining part of the function. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index bd85c98..1341f327 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -243,17 +243,17 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); break; default: /* configure a full-word SYNC pulse (LRCLK) */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); /* make 1st data bit occur one ACLK cycle after the frame sync */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); break; } -- cgit v0.10.2 From 487dce8823cdcb70e645e5312a0d4f7081e1ad13 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:31 +0200 Subject: ASoC: davinci-mcasp: Simplify FIFO configuration code The FIFO registers base address is different in dm646x compared to newer SoCs with McASP IP. Instead of using two paths (switch/case) to handle the difference we can simply pick the correct base address beforehand and use offsets to address the register we need to configure. With this change the indentation depth can be reduced as well. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1341f327..72ea458 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -38,6 +38,7 @@ struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; + u32 fifo_base; struct device *dev; /* McASP specific data */ @@ -153,38 +154,20 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { + u32 reg; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* enable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - break; - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + - DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_start_tx(mcasp); } else { if (mcasp->rxnumevt) { /* enable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - break; - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + - DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_start_rx(mcasp); } @@ -204,31 +187,18 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { + u32 reg; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* disable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - break; - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_stop_tx(mcasp); } else { if (mcasp->rxnumevt) { /* disable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - break; - - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_stop_rx(mcasp); } @@ -438,6 +408,7 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, u8 ser; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; + u32 reg; /* Default configuration */ mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -488,37 +459,20 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, if (mcasp->txnumevt * tx_ser > 64) mcasp->txnumevt = 1; - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, tx_ser, - NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); - break; - default: - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, - tx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, - ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); - } + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_mod_bits(mcasp->base + reg, tx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp->base + reg, + ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); } if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (mcasp->rxnumevt * rx_ser > 64) mcasp->rxnumevt = 1; - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, rx_ser, - NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); - break; - default: - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, - rx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, - ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); - } + + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_mod_bits(mcasp->base + reg, rx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp->base + reg, + ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } return 0; @@ -974,6 +928,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; + if (mcasp->version < MCASP_VERSION_3) + mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; + else + mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; + mcasp->dev = &pdev->dev; dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 80e5a18..8fed757 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -90,14 +90,14 @@ #define DAVINCI_MCASP_RXBUF_REG 0x280 /* McASP FIFO Registers */ -#define DAVINCI_MCASP_WFIFOCTL (0x1010) -#define DAVINCI_MCASP_WFIFOSTS (0x1014) -#define DAVINCI_MCASP_RFIFOCTL (0x1018) -#define DAVINCI_MCASP_RFIFOSTS (0x101C) -#define MCASP_VER3_WFIFOCTL (0x1000) -#define MCASP_VER3_WFIFOSTS (0x1004) -#define MCASP_VER3_RFIFOCTL (0x1008) -#define MCASP_VER3_RFIFOSTS (0x100C) +#define DAVINCI_MCASP_V2_AFIFO_BASE (0x1010) +#define DAVINCI_MCASP_V3_AFIFO_BASE (0x1000) + +/* FIFO register offsets from AFIFO base */ +#define MCASP_WFIFOCTL_OFFSET (0x0) +#define MCASP_WFIFOSTS_OFFSET (0x4) +#define MCASP_RFIFOCTL_OFFSET (0x8) +#define MCASP_RFIFOSTS_OFFSET (0xc) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management -- cgit v0.10.2 From cbc7956c81eea644c0d99aee43f1632897703300 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:32 +0200 Subject: ASoC: davinci-mcasp: Data source (bus) selection support The audio data to/from McASP can be sent/received via two method: Via the data port (preferred) or via the configuration bus. Currently the driver assumes that all data communication will be done via the data port. This patch adds support for selecting the configuration port as data interface. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 72ea458..35a6292 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -53,6 +53,8 @@ struct davinci_mcasp { u8 txnumevt; u8 rxnumevt; + bool dat_port; + #ifdef CONFIG_PM_SLEEP struct { u32 txfmtctl; @@ -482,6 +484,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; + u32 busel = 0; active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) @@ -489,11 +492,15 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + if (!mcasp->dat_port) + busel = TXSEL; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, + busel | TXORD); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, @@ -504,7 +511,8 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, + busel | RXORD); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) @@ -928,23 +936,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; - if (mcasp->version < MCASP_VERSION_3) - mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; - else - mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; mcasp->dev = &pdev->dev; dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); - if (!dat) - dat = mem; + if (dat) + mcasp->dat_port = true; dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; - dma_data->dma_addr = dat->start + pdata->tx_dma_offset; + if (dat) + dma_data->dma_addr = dat->start; + else + dma_data->dma_addr = mem->start + pdata->tx_dma_offset; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) @@ -957,7 +964,18 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; - dma_data->dma_addr = dat->start + pdata->rx_dma_offset; + if (dat) + dma_data->dma_addr = dat->start; + else + dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + + if (mcasp->version < MCASP_VERSION_3) { + mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; + /* dma_data->dma_addr is pointing to the data port address */ + mcasp->dat_port = true; + } else { + mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; + } res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) -- cgit v0.10.2 From 4dcb5a0bffaa7dc51e738c4f651a31993b1eb08b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:33 +0200 Subject: ASoC: davinci-mcasp: Fix synchronous master receive mode In synchronous mode both transmit and receive sections are using the TX clocks. In setup like this the TX clocks need to be enabled when capture is running. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 35a6292..93f2e29 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -48,6 +48,7 @@ struct davinci_mcasp { u8 *serial_dir; u8 version; u16 bclk_lrclk_ratio; + int streams; /* McASP FIFO related */ u8 txnumevt; @@ -110,10 +111,31 @@ static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) printk(KERN_ERR "GBLCTL write error\n"); } +static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) +{ + u32 rxfmctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG); + u32 aclkxctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG); + + return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE; +} + static void mcasp_start_rx(struct davinci_mcasp *mcasp) { mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + + /* + * When ASYNC == 0 the transmit and receive sections operate + * synchronously from the transmit clock and frame sync. We need to make + * sure that the TX signlas are enabled when starting reception. + */ + if (mcasp_is_synchronous(mcasp)) { + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, + TXHCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, + TXCLKRST); + } + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); @@ -123,6 +145,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + + if (mcasp_is_synchronous(mcasp)) + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, + TXFSRST); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -158,6 +184,8 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { u32 reg; + mcasp->streams++; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* enable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; @@ -177,13 +205,29 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* + * In synchronous mode stop the TX clocks if no other stream is + * running + */ + if (mcasp_is_synchronous(mcasp) && !mcasp->streams) + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); } static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + u32 val = 0; + + /* + * In synchronous mode keep TX clocks running if the capture stream is + * still running. + */ + if (mcasp_is_synchronous(mcasp) && mcasp->streams) + val = TXHCLKRST | TXCLKRST | TXFSRST; + + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, val); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); } @@ -191,6 +235,8 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { u32 reg; + mcasp->streams--; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* disable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; -- cgit v0.10.2 From 453c499028bf2ecf3b31ccb7c3657fe1b0b28943 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:34 +0200 Subject: ASoC: davinci-mcasp: Support for McASP version found in DRA7xx The IP in DRA7xx is similar to the IP found in TI81xxAM3xxx/AM4xxx type of SoCs but it is is integrated with sDMA instead of eDMA. The suitable pcm driver for DRA7xx is the omap-pcm driver which is using dmaengine. In the driver we can configure both dma related structures used for eDMA and sDMA. The only thing we need to make sure that we set the correct dma_data at startup with snd_soc_dai_set_dma_data() Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 1eed972..990fa71 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -5,6 +5,7 @@ Required properties: "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) + "ti,dra7-mcasp-audio" : for DRA7xx platforms - reg : Should contain reg specifiers for the entries in the reg-names property. - reg-names : Should contain: diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 689a856..5245992 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -92,6 +92,7 @@ enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ MCASP_VERSION_3, /* TI81xx/AM33xx */ + MCASP_VERSION_4, /* DRA7xxx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 93f2e29..fc8c13d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -31,12 +31,14 @@ #include #include #include +#include #include "davinci-pcm.h" #include "davinci-mcasp.h" struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; struct device *dev; @@ -458,7 +460,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, u8 max_active_serializers = (channels + slots - 1) / slots; u32 reg; /* Default configuration */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + if (mcasp->version != MCASP_VERSION_4) + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, + MCASP_SOFT); /* All PINS as McASP */ mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); @@ -605,6 +609,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data = + &mcasp->dma_data[substream->stream]; int word_length; u8 fifo_level; u8 slots = mcasp->tdm_slots; @@ -666,6 +672,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_data->maxburst = fifo_level; + davinci_config_channel_size(mcasp, word_length); return 0; @@ -711,7 +719,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params); + if (mcasp->version == MCASP_VERSION_4) + snd_soc_dai_set_dma_data(dai, substream, + &mcasp->dma_data[substream->stream]); + else + snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params); + return 0; } @@ -794,6 +807,13 @@ static struct snd_platform_data omap2_mcasp_pdata = { .version = MCASP_VERSION_3, }; +static struct snd_platform_data dra7_mcasp_pdata = { + .tx_dma_offset = 0x200, + .rx_dma_offset = 0x284, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_4, +}; + static const struct of_device_id mcasp_dt_ids[] = { { .compatible = "ti,dm646x-mcasp-audio", @@ -807,6 +827,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,am33xx-mcasp-audio", .data = &omap2_mcasp_pdata, }, + { + .compatible = "ti,dra7-mcasp-audio", + .data = &dra7_mcasp_pdata, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); @@ -999,6 +1023,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->dma_addr = mem->start + pdata->tx_dma_offset; + /* Unconditional dmaengine stuff */ + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr; + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) dma_data->channel = res->start; @@ -1015,6 +1042,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + /* Unconditional dmaengine stuff */ + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr; + if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; /* dma_data->dma_addr is pointing to the data port address */ @@ -1029,6 +1059,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->channel = pdata->rx_dma_channel; + /* Unconditional dmaengine stuff */ + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx"; + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx"; + dev_set_drvdata(&pdev->dev, mcasp); ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); @@ -1036,10 +1070,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto err_unregister_component; + if (mcasp->version != MCASP_VERSION_4) { + ret = davinci_soc_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + goto err_unregister_component; + } } return 0; @@ -1054,9 +1090,11 @@ err_release_clk: static int davinci_mcasp_remove(struct platform_device *pdev) { + struct davinci_mcasp *mcasp = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - davinci_soc_platform_unregister(&pdev->dev); + if (mcasp->version != MCASP_VERSION_4) + davinci_soc_platform_unregister(&pdev->dev); pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- cgit v0.10.2 From f68205a7f8c0b1fd02cec6116bbb66bb4fd7bc51 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:36 +0200 Subject: ASoC: davinci-mcasp: Change IO functions parameter list Instead of passing __iomem address (mcasp->base + register_offset) pass the main mcasp structure and only access the mcasp->base in the low level IO functions. In most cases this helps with code readability and it will make it easier to switch over to regmap in the future. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index fc8c13d..19c6662 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -71,60 +71,67 @@ struct davinci_mcasp { #endif }; -static inline void mcasp_set_bits(void __iomem *reg, u32 val) +static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, + u32 val) { + void __iomem *reg = mcasp->base + offset; __raw_writel(__raw_readl(reg) | val, reg); } -static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +static inline void mcasp_clr_bits(struct davinci_mcasp *mcasp, u32 offset, + u32 val) { + void __iomem *reg = mcasp->base + offset; __raw_writel((__raw_readl(reg) & ~(val)), reg); } -static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +static inline void mcasp_mod_bits(struct davinci_mcasp *mcasp, u32 offset, + u32 val, u32 mask) { + void __iomem *reg = mcasp->base + offset; __raw_writel((__raw_readl(reg) & ~mask) | val, reg); } -static inline void mcasp_set_reg(void __iomem *reg, u32 val) +static inline void mcasp_set_reg(struct davinci_mcasp *mcasp, u32 offset, + u32 val) { - __raw_writel(val, reg); + __raw_writel(val, mcasp->base + offset); } -static inline u32 mcasp_get_reg(void __iomem *reg) +static inline u32 mcasp_get_reg(struct davinci_mcasp *mcasp, u32 offset) { - return (unsigned int)__raw_readl(reg); + return (u32)__raw_readl(mcasp->base + offset); } -static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +static void mcasp_set_ctl_reg(struct davinci_mcasp *mcasp, u32 ctl_reg, u32 val) { int i = 0; - mcasp_set_bits(regs, val); + mcasp_set_bits(mcasp, ctl_reg, val); /* programming GBLCTL needs to read back from GBLCTL and verfiy */ /* loop count is to avoid the lock-up */ for (i = 0; i < 1000; i++) { - if ((mcasp_get_reg(regs) & val) == val) + if ((mcasp_get_reg(mcasp, ctl_reg) & val) == val) break; } - if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + if (i == 1000 && ((mcasp_get_reg(mcasp, ctl_reg) & val) != val)) printk(KERN_ERR "GBLCTL write error\n"); } static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) { - u32 rxfmctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG); - u32 aclkxctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG); + u32 rxfmctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); + u32 aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE; } static void mcasp_start_rx(struct davinci_mcasp *mcasp) { - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); /* * When ASYNC == 0 the transmit and receive sections operate @@ -132,25 +139,22 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) * sure that the TX signlas are enabled when starting reception. */ if (mcasp_is_synchronous(mcasp)) { - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, - TXHCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, - TXCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); } - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, - TXFSRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -158,14 +162,14 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) u8 offset = 0, i; u32 cnt; - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); for (i = 0; i < mcasp->num_serializer; i++) { if (mcasp->serial_dir[i] == TX_MODE) { offset = i; @@ -175,11 +179,11 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for TX ready */ cnt = 0; - while (!(mcasp_get_reg(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(offset)) & TXSTATE) && (cnt < 100000)) cnt++; - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -191,15 +195,15 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* enable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + mcasp_set_bits(mcasp, reg, FIFO_ENABLE); } mcasp_start_tx(mcasp); } else { if (mcasp->rxnumevt) { /* enable FIFO */ reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + mcasp_set_bits(mcasp, reg, FIFO_ENABLE); } mcasp_start_rx(mcasp); } @@ -212,10 +216,10 @@ static void mcasp_stop_rx(struct davinci_mcasp *mcasp) * running */ if (mcasp_is_synchronous(mcasp) && !mcasp->streams) - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, 0); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); } static void mcasp_stop_tx(struct davinci_mcasp *mcasp) @@ -229,8 +233,8 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) if (mcasp_is_synchronous(mcasp) && mcasp->streams) val = TXHCLKRST | TXCLKRST | TXFSRST; - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, val); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); } static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) @@ -242,13 +246,13 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* disable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); } mcasp_stop_tx(mcasp); } else { if (mcasp->rxnumevt) { /* disable FIFO */ reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); } mcasp_stop_rx(mcasp); } @@ -258,62 +262,57 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - void __iomem *base = mcasp->base; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); break; default: /* configure a full-word SYNC pulse (LRCLK) */ - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); /* make 1st data bit occur one ACLK cycle after the frame sync */ - mcasp_set_bits(base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* codec is clock and frame slave */ - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | ACLKR); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - AFSX | AFSR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | ACLKR); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - AFSX | AFSR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, + ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); break; default: @@ -322,35 +321,35 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_NF: - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; case SND_SOC_DAIFMT_NB_IF: - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; case SND_SOC_DAIFMT_IB_IF: - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; case SND_SOC_DAIFMT_NB_NF: - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; default: @@ -366,16 +365,16 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div switch (div_id) { case 0: /* MCLK divider */ - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; case 1: /* BCLK divider */ - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKRCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); break; @@ -396,13 +395,13 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (dir == SND_SOC_CLOCK_OUT) { - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } return 0; @@ -432,19 +431,18 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, fmt = (word_length >> 1) - 1; if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(tx_rotate), TXROT(7)); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rx_rotate), RXROT(7)); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXMASK_REG, - mask); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), + RXSSZ(0x0F)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), + TXSSZ(0x0F)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), + TXROT(7)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), + RXROT(7)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask); } - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask); return 0; } @@ -461,38 +459,33 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, u32 reg; /* Default configuration */ if (mcasp->version != MCASP_VERSION_4) - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, - MCASP_SOFT); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); /* All PINS as McASP */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + mcasp_set_reg(mcasp, DAVINCI_MCASP_PFUNC_REG, 0x00000000); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, - TXDATADMADIS); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); } else { - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_REVTCTL_REG, - RXDATADMADIS); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS); } for (i = 0; i < mcasp->num_serializer; i++) { - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), - mcasp->serial_dir[i]); + mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp->serial_dir[i]); if (mcasp->serial_dir[i] == TX_MODE && tx_ser < max_active_serializers) { - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, - AXR(i)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; } else if (mcasp->serial_dir[i] == RX_MODE && rx_ser < max_active_serializers) { - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, - AXR(i)); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; } else { - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), - SRMOD_INACTIVE, SRMOD_MASK); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -512,9 +505,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, mcasp->txnumevt = 1; reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_mod_bits(mcasp->base + reg, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + reg, - ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp, reg, tx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp, reg, ((mcasp->txnumevt * tx_ser) << 8), + NUMEVT_MASK); } if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -522,9 +515,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, mcasp->rxnumevt = 1; reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_mod_bits(mcasp->base + reg, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + reg, - ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp, reg, rx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp, reg, ((mcasp->rxnumevt * rx_ser) << 8), + NUMEVT_MASK); } return 0; @@ -540,7 +533,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) for (i = 0; i < active_slots; i++) mask |= (1 << i); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); if (!mcasp->dat_port) busel = TXSEL; @@ -548,26 +541,24 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - busel | TXORD); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); else printk(KERN_ERR "playback tdm slot %d not supported\n", mcasp->tdm_slots); } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, - busel | RXORD); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); else printk(KERN_ERR "capture tdm slot %d not supported\n", mcasp->tdm_slots); @@ -579,27 +570,24 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(6) | TXSSZ(15)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, - AFSXE | FSXMOD(0x180)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); /* Set the TX tdm : for all the slots */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); /* Set the TX clock controls : div = 1 and internal */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, - ACLKXE | TX_ASYNC); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE | TX_ASYNC); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); /* Enable the DIT */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -1106,15 +1094,14 @@ static int davinci_mcasp_remove(struct platform_device *pdev) static int davinci_mcasp_suspend(struct device *dev) { struct davinci_mcasp *mcasp = dev_get_drvdata(dev); - void __iomem *base = mcasp->base; - mcasp->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); - mcasp->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); - mcasp->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); - mcasp->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); - mcasp->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); - mcasp->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); - mcasp->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + mcasp->context.txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); + mcasp->context.rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); + mcasp->context.txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); + mcasp->context.rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); + mcasp->context.aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); + mcasp->context.aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); + mcasp->context.pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); return 0; } @@ -1122,15 +1109,14 @@ static int davinci_mcasp_suspend(struct device *dev) static int davinci_mcasp_resume(struct device *dev) { struct davinci_mcasp *mcasp = dev_get_drvdata(dev); - void __iomem *base = mcasp->base; - - mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); - mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); + mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); return 0; } -- cgit v0.10.2 From b14899da9ddeb8501db13fd08d0d1a8af61529c5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:37 +0200 Subject: ASoC: davinci-mcasp: Correct am33xx snd_platform_data name An earlier patch overlooked this when the compatible has been changed from omap2 -> am33x. Rename omap2_mcasp_pdata to am33xx_mcasp_pdata. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 19c6662..8ec8795 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -788,7 +788,7 @@ static struct snd_platform_data da830_mcasp_pdata = { .version = MCASP_VERSION_2, }; -static struct snd_platform_data omap2_mcasp_pdata = { +static struct snd_platform_data am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, .asp_chan_q = EVENTQ_0, @@ -813,7 +813,7 @@ static const struct of_device_id mcasp_dt_ids[] = { }, { .compatible = "ti,am33xx-mcasp-audio", - .data = &omap2_mcasp_pdata, + .data = &am33xx_mcasp_pdata, }, { .compatible = "ti,dra7-mcasp-audio", -- cgit v0.10.2 From ae726e93946403b14f8cad20d5cbd22d015c9106 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:35 +0200 Subject: ASoC: davinci-mcasp: Support for fck reparenting Optional DT property to specify the desired parent clock for the McASP fck clock. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 990fa71..569b26c 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -37,7 +37,8 @@ Optional properties: - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details please refer to pinctrl-bindings.txt - +- fck_parent : Should contain a valid clock name which will be used as parent + for the McASP fck Example: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8ec8795..b7858bf 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -823,6 +824,46 @@ static const struct of_device_id mcasp_dt_ids[] = { }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); +static int mcasp_reparent_fck(struct platform_device *pdev) +{ + struct device_node *node = pdev->dev.of_node; + struct clk *gfclk, *parent_clk; + const char *parent_name; + int ret; + + if (!node) + return 0; + + parent_name = of_get_property(node, "fck_parent", NULL); + if (!parent_name) + return 0; + + gfclk = clk_get(&pdev->dev, "fck"); + if (IS_ERR(gfclk)) { + dev_err(&pdev->dev, "failed to get fck\n"); + return PTR_ERR(gfclk); + } + + parent_clk = clk_get(NULL, parent_name); + if (IS_ERR(parent_clk)) { + dev_err(&pdev->dev, "failed to get parent clock\n"); + ret = PTR_ERR(parent_clk); + goto err1; + } + + ret = clk_set_parent(gfclk, parent_clk); + if (ret) { + dev_err(&pdev->dev, "failed to reparent fck\n"); + goto err2; + } + +err2: + clk_put(parent_clk); +err1: + clk_put(gfclk); + return ret; +} + static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct platform_device *pdev) { @@ -1052,6 +1093,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx"; dev_set_drvdata(&pdev->dev, mcasp); + + mcasp_reparent_fck(pdev); + ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); -- cgit v0.10.2 From a073278228836d7d18fdd6c40b619919c0befb64 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2013 20:46:59 -0800 Subject: ASoC: fsi: remove original filter from fsi_dma_probe() Remove original filter from fsi_dma_probe(), and use SH-DMA suitable filter. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b33ca7c..6101055 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -232,9 +232,9 @@ struct fsi_stream { * these are for DMAEngine */ struct dma_chan *chan; - struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int dma_id; int loop_cnt; int additional_pos; }; @@ -1410,15 +1410,6 @@ static void fsi_dma_do_work(struct work_struct *work) } } -static bool fsi_dma_filter(struct dma_chan *chan, void *param) -{ - struct sh_dmae_slave *slave = param; - - chan->private = slave; - - return true; -} - static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { schedule_work(&io->work); @@ -1446,15 +1437,34 @@ static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; + int is_play = fsi_stream_is_play(fsi, io); dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); + io->chan = dma_request_slave_channel_compat(mask, + shdma_chan_filter, (void *)io->dma_id, + dev, is_play ? "tx" : "rx"); + if (io->chan) { + struct dma_slave_config cfg; + int ret; + + cfg.slave_id = io->dma_id; + cfg.dst_addr = 0; /* use default addr */ + cfg.src_addr = 0; /* use default addr */ + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + ret = dmaengine_slave_config(io->chan, &cfg); + if (ret < 0) { + dma_release_channel(io->chan); + io->chan = NULL; + } + } + if (!io->chan) { /* switch to PIO handler */ - if (fsi_stream_is_play(fsi, io)) + if (is_play) fsi->playback.handler = &fsi_pio_push_handler; else fsi->capture.handler = &fsi_pio_pop_handler; @@ -1960,7 +1970,7 @@ static void fsi_handler_init(struct fsi_priv *fsi, fsi->capture.priv = fsi; if (info->tx_id) { - fsi->playback.slave.shdma_slave.slave_id = info->tx_id; + fsi->playback.dma_id = info->tx_id; fsi->playback.handler = &fsi_dma_push_handler; } } -- cgit v0.10.2 From 3688569e8173e84cd95d98f158245e17bca4f593 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 15:23:15 +0100 Subject: ASoC: samsung: Provide helper for DMA init In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23..4a88e36 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4..eab0050 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- cgit v0.10.2 From d37bdf736d9b7a198d35aaaf611e96ddc2e00ddf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Dec 2013 14:14:52 +0000 Subject: ASoC: samsung: Use ASoC dmaengine code where possible Since all Exynos platforms have been converted to dmaengine and many of the older platforms are in the process of conversion they do not need to use the legacy s3c-dma APIs for DMA but can instead use the standard ASoC dmaengine helpers. This both allows them to benefit from improvements implemented in the generic code and supports multiplatform. This patch includes some fixes from Padma for Exynos SoCs, her testing was on a slightly earlier version of the patch due to unrelated breakage preventing testing. Signed-off-by: Mark Brown Tested By: Padmavathi Venna diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 37459df..27930fc 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,13 +1,22 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C64XX_DMA if ARCH_S3C64XX - select S3C24XX_DMA if ARCH_S3C24XX + select S3C2410_DMA if ARCH_S3C24XX + select S3C64XX_PL080 if ARCH_S3C64XX + select SND_S3C_DMA if !ARCH_S3C24XX + select SND_S3C_DMA_LEGACY if ARCH_S3C24XX + select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. +config SND_S3C_DMA + tristate + +config SND_S3C_DMA_LEGACY + tristate + config SND_S3C24XX_I2S tristate select S3C2410_DMA diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 709f605..86715d8 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := dma.o +snd-soc-s3c-dma-objs := dmaengine.o +snd-soc-s3c-dma-legacy-objs := dma.o snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o @@ -9,7 +10,8 @@ snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o -obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c24xx.o +obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o +obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index fb09a1c..225e537 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -12,6 +12,8 @@ #ifndef _S3C_AUDIO_H #define _S3C_AUDIO_H +#include + struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ @@ -20,6 +22,7 @@ struct s3c_dma_params { unsigned ch; struct samsung_dma_ops *ops; char *ch_name; + struct snd_dmaengine_dai_dma_data dma_data; }; void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c new file mode 100644 index 0000000..3be479d --- /dev/null +++ b/sound/soc/samsung/dmaengine.c @@ -0,0 +1,84 @@ +/* + * dmaengine.c - Samsung dmaengine wrapper + * + * Author: Mark Brown + * Copyright 2013 Linaro + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "dma.h" + +#ifdef CONFIG_ARCH_S3C64XX +#define filter_fn pl08x_filter_id +#else +#define filter_fn NULL +#endif + +static const struct snd_dmaengine_pcm_config samsung_dmaengine_pcm_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = filter_fn, +}; + +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + struct snd_dmaengine_dai_dma_data *playback_data = NULL; + struct snd_dmaengine_dai_dma_data *capture_data = NULL; + + if (playback) { + playback_data = &playback->dma_data; + playback_data->filter_data = (void *)playback->channel; + playback_data->chan_name = playback->ch_name; + playback_data->addr = playback->dma_addr; + playback_data->addr_width = playback->dma_size; + } + if (capture) { + capture_data = &capture->dma_data; + capture_data->filter_data = (void *)capture->channel; + capture_data->chan_name = capture->ch_name; + capture_data->addr = capture->dma_addr; + capture_data->addr_width = capture->dma_size; + } + + snd_soc_dai_init_dma_data(dai, playback_data, capture_data); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + +int samsung_asoc_dma_platform_register(struct device *dev) +{ + return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME | + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); + +void samsung_asoc_dma_platform_unregister(struct device *dev) +{ + return snd_dmaengine_pcm_unregister(dev); +} +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); + +MODULE_AUTHOR("Mark Brown "); +MODULE_DESCRIPTION("Samsung dmaengine ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index eab0050..92f64363 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -702,6 +702,8 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } writel(mod, i2s->addr + I2SMOD); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + i2s->frmclk = params_rate(params); return 0; -- cgit v0.10.2 From 753834cb593da03b4efc468bb8cb76dbc0743b31 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Sat, 14 Dec 2013 13:29:11 +0800 Subject: ASoC: tegra20-ac97: add missing clk_disable_unprepare() on error path Add the missing clk_disable_unprepare() before return from tegra20_ac97_platform_probe() in the error handling case. Signed-off-by: Wei Yongjun Reviewed-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index ae27bcd..088518d 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -404,7 +404,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); if (ret) { dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, @@ -412,7 +412,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -428,6 +428,8 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_clk_disable_unprepare: + clk_disable_unprepare(ac97->clk_ac97); err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: -- cgit v0.10.2 From 252e91ff1094eefacd25b401c3b77e549803cae6 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 13 Dec 2013 14:43:02 +0800 Subject: ASoC: sgtl5000: read chip revision for once Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe() reads it out from register, so that we can use it in sgtl5000_enable_regulators() with no need to read register again. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..bd291d2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -115,6 +115,7 @@ struct sgtl5000_priv { struct ldo_regulator *ldo; struct regmap *regmap; struct clk *mclk; + int revision; }; /* @@ -1300,9 +1301,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; int ret; - int rev; int i; int external_vddd = 0; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); @@ -1332,14 +1331,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) * workaround for revision 0x11 and later, * roll back to use internal LDO */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); - if (ret) - goto err_regulator_disable; - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - - if (external_vddd && rev >= 0x11) { + if (external_vddd && sgtl5000->revision >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1362,9 +1354,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return 0; -err_regulator_disable: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); err_regulator_free: regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1566,6 +1555,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + sgtl5000->revision = rev; i2c_set_clientdata(client, sgtl5000); -- cgit v0.10.2 From 7fd7a48bd584a66cc4b0e3b92bb75b061578e19e Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Mon, 16 Dec 2013 13:07:23 +0100 Subject: ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap() devm_request_and_ioremap() has been deprecated in favour of devm_ioremap_resource(). Fixes the following coccinelle warning: sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195 Generated by: coccinelle/api/devm_ioremap_resource.cocci Signed-off-by: Fengguang Wu Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 0822c77..7f91a86 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -192,9 +192,9 @@ static int axi_i2s_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, res); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base, &axi_i2s_regmap_config); -- cgit v0.10.2 From bbe580302d33cff282129e26c44f9c3450d6a086 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Mon, 16 Dec 2013 13:07:24 +0100 Subject: ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap() devm_request_and_ioremap() has been deprecated in favour of devm_ioremap_resource(). Fixes the following coccinelle warning: sound/soc/adi/axi-spdif.c:194:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 194 Generated by: coccinelle/api/devm_ioremap_resource.cocci Signed-off-by: Fengguang Wu Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index d5408d2..8db7a99 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -191,9 +191,9 @@ static int axi_spdif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, spdif); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, res); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); spdif->regmap = devm_regmap_init_mmio(&pdev->dev, base, &axi_spdif_regmap_config); -- cgit v0.10.2 From 3c43c69537daa044c61965fad24e24ad392c4166 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:49:22 +0000 Subject: ASoC: arizona: Use async writes Where possible write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Tested-by: Charles Keepax Reviewed-by: Charles Keepax diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eb9f5d4..6bfd803 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -93,7 +93,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (!priv->spk_ena && manual_ena) { - snd_soc_write(codec, 0x4f5, 0x25a); + regmap_write_async(arizona->regmap, 0x4f5, 0x25a); priv->spk_ena_pending = true; } break; @@ -105,12 +105,13 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, return -EBUSY; } - snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, - 1 << w->shift, 1 << w->shift); + regmap_update_bits_async(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 1 << w->shift); if (priv->spk_ena_pending) { msleep(75); - snd_soc_write(codec, 0x4f5, 0xda); + regmap_write_async(arizona->regmap, 0x4f5, 0xda); priv->spk_ena_pending = false; priv->spk_ena++; } @@ -119,16 +120,19 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, if (manual_ena) { priv->spk_ena--; if (!priv->spk_ena) - snd_soc_write(codec, 0x4f5, 0x25a); + regmap_write_async(arizona->regmap, + 0x4f5, 0x25a); } - snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, - 1 << w->shift, 0); + regmap_update_bits_async(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 0); break; case SND_SOC_DAPM_POST_PMD: if (manual_ena) { if (!priv->spk_ena) - snd_soc_write(codec, 0x4f5, 0x0da); + regmap_write_async(arizona->regmap, + 0x4f5, 0x0da); } break; } @@ -687,6 +691,7 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, int event) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct arizona *arizona = priv->arizona; unsigned int mask = 1 << w->shift; unsigned int val; @@ -709,7 +714,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, if (priv->arizona->hpdet_magic) val = 0; - snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val); + regmap_update_bits_async(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1, + mask, val); return arizona_out_ev(w, kcontrol, event); } @@ -864,6 +870,8 @@ EXPORT_SYMBOL_GPL(arizona_set_sysclk); static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; int lrclk, bclk, mode, base; base = dai->driver->base; @@ -920,17 +928,19 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, - bclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, - ARIZONA_AIF1TX_LRCLK_INV | - ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, - ARIZONA_AIF1RX_LRCLK_INV | - ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, - ARIZONA_AIF1_FMT_MASK, mode); + regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | + ARIZONA_AIF1_BCLK_MSTR, + bclk); + regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); return 0; } @@ -1182,18 +1192,22 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, if (ret != 0) return ret; - snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, - ARIZONA_AIF1TX_BCPF_MASK, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, - ARIZONA_AIF1RX_BCPF_MASK, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, - ARIZONA_AIF1TX_WL_MASK | - ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, - ARIZONA_AIF1RX_WL_MASK | - ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); return 0; } @@ -1446,31 +1460,31 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, struct arizona_fll_cfg *cfg, int source, bool sync) { - regmap_update_bits(arizona->regmap, base + 3, - ARIZONA_FLL1_THETA_MASK, cfg->theta); - regmap_update_bits(arizona->regmap, base + 4, - ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); - regmap_update_bits(arizona->regmap, base + 5, - ARIZONA_FLL1_FRATIO_MASK, - cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); - regmap_update_bits(arizona->regmap, base + 6, - ARIZONA_FLL1_CLK_REF_DIV_MASK | - ARIZONA_FLL1_CLK_REF_SRC_MASK, - cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | - source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits_async(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits_async(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); if (sync) - regmap_update_bits(arizona->regmap, base + 0x7, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); else - regmap_update_bits(arizona->regmap, base + 0x9, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); - regmap_update_bits(arizona->regmap, base + 2, - ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, - ARIZONA_FLL1_CTRL_UPD | cfg->n); + regmap_update_bits_async(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); } static bool arizona_is_enabled_fll(struct arizona_fll *fll) @@ -1503,9 +1517,9 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits_async(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); @@ -1515,15 +1529,15 @@ static void arizona_enable_fll(struct arizona_fll *fll, use_sync = true; } } else if (fll->sync_src >= 0) { - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits_async(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, sync, fll->sync_src, false); - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); return; @@ -1534,11 +1548,12 @@ static void arizona_enable_fll(struct arizona_fll *fll, * sync source. */ if (use_sync && fll->sync_freq > 100000) - regmap_update_bits(arizona->regmap, fll->base + 0x17, - ARIZONA_FLL1_SYNC_BW, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); else - regmap_update_bits(arizona->regmap, fll->base + 0x17, - ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + regmap_update_bits_async(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, + ARIZONA_FLL1_SYNC_BW); if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); @@ -1546,14 +1561,14 @@ static void arizona_enable_fll(struct arizona_fll *fll, /* Clear any pending completions */ try_wait_for_completion(&fll->ok); - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, 0); - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, - ARIZONA_FLL1_SYNC_ENA); + regmap_update_bits_async(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); ret = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); @@ -1566,8 +1581,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, -- cgit v0.10.2 From bd4893492bfc6081da55e445d200c1a832770a06 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:49:50 +0000 Subject: ASoC: wm5102: Use async writes When writing the patch write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Reviewed-by: Charles Keepax diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a08e8bf..ce9c8e1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -601,8 +601,8 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (patch) for (i = 0; i < patch_size; i++) - regmap_write(regmap, patch[i].reg, - patch[i].def); + regmap_write_async(regmap, patch[i].reg, + patch[i].def); break; default: -- cgit v0.10.2 From 959e4083cd9c43fb3e818984926f9c590ee0aa2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:49:56 +0000 Subject: ASoC: wm5110: Use async writes When writing the patch write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Tested-by: Charles Keepax Reviewed-by: Charles Keepax diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0e63d8c..ebcbe78 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -105,8 +105,8 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (patch) for (i = 0; i < patch_size; i++) - regmap_write(regmap, patch[i].reg, - patch[i].def); + regmap_write_async(regmap, patch[i].reg, + patch[i].def); break; default: -- cgit v0.10.2 From 1f4fe272f068813377845a959cab5ce786a155bf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:50:04 +0000 Subject: ASoC: wm8997: Use async writes When writing the patch write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Reviewed-by: Charles Keepax diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 1392bb3..555115e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -103,8 +103,8 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (patch) for (i = 0; i < patch_size; i++) - regmap_write(regmap, patch[i].reg, - patch[i].def); + regmap_write_async(regmap, patch[i].reg, + patch[i].def); break; default: break; -- cgit v0.10.2 From 48b752ac2f80f483a3059ae109f9de02dcc054dd Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Tue, 17 Dec 2013 16:22:24 +0800 Subject: ASoC: mmp-pcm: config pcm slave via generic dmaengine use snd_dmaengine_pcm_prepare_slave_config to set slave config, and remove the max_burst_size = 4 hard code. select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm. Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4db74a0..6473052 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,7 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM help Say Y if you want to add support for codecs attached to diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 7929e19..682ee52 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -67,27 +67,15 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config slave_config; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!dma_params) - return 0; - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + ret = + snd_dmaengine_pcm_prepare_slave_config(substream, params, + &slave_config); if (ret) return ret; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->addr; - slave_config.dst_maxburst = 4; - } else { - slave_config.src_addr = dma_params->addr; - slave_config.src_maxburst = 4; - } - ret = dmaengine_slave_config(chan, &slave_config); if (ret) return ret; -- cgit v0.10.2 From 58e354be337dff43dbb66c4564bb9479354cc5dd Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Tue, 17 Dec 2013 11:52:43 +0100 Subject: ASoC: tegra: Tweak matching of AC97 components Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach Acked-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e11963..45b5789 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- cgit v0.10.2 From 5095f55d7cc327026daaa3fa583aa4c1388ca556 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 18 Dec 2013 09:25:48 +0000 Subject: ASoC: wm_adsp: Remove duplicate info message for DSP RAM ready Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6b1c01c..8f720de 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1488,7 +1488,6 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) } adsp_dbg(dsp, "RAM ready after %d polls\n", count); - adsp_info(dsp, "RAM ready after %d polls\n", count); return 0; } -- cgit v0.10.2 From dd407a3243234c6a17ba624d698e6824067003c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Dec 2013 13:50:10 +0000 Subject: ASoC: fsl/mxs: Remove unnecessarily gendered language The kernel as a number of cases of gendered language. The majority of these refer to objects that don't have gender in English, and so I've replaced them with "it" and "its". Some refer to people (developers or users), and I've replaced these with the singular "they" variant. Some are simply typos that I've fixed up. I've left cases where gendered language was used to refer to specific individuals, was a quote or is part of license text. Signed-off-by: Matthew Garrett Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index fb9bb9e..d570f8c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -852,7 +852,7 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) } /** - * find_ssi_node -- returns the SSI node that points to his DMA channel node + * find_ssi_node -- returns the SSI node that points to its DMA channel node * * Although this DMA driver attempts to operate independently of the other * devices, it still needs to determine some information about the SSI device diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54e622a..d6cb9a5 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -50,9 +50,9 @@ static struct mxs_saif *mxs_saif[2]; * This also means that both SAIFs must operate at the same sample rate. * * We abstract this as each saif has a master, the master could be - * himself or other saifs. In the generic saif driver, saif does not need - * to know the different clkmux. Saif only needs to know who is his master - * and operating his master to generate the proper clock rate for him. + * itself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is its master + * and operating its master to generate the proper clock rate for it. * The master id is provided in mach-specific layer according to different * clkmux setting. */ @@ -76,7 +76,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK * is provided by other SAIF, we provide a interface here to get its master * from its master_id. - * Note that the master could be himself. + * Note that the master could be itself. */ static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) { @@ -516,7 +516,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } /* - * If the saif's master is not himself, we also need to enable + * If the saif's master is not itself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { -- cgit v0.10.2 From 4355082149429d1f87b6fbfc3ebc6305a5372ce2 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 17 Dec 2013 11:24:38 +0800 Subject: ASoC: Add SAI SoC Digital Audio Interface driver. This adds Freescale SAI ASoC Audio support. This implementation is only compatible with device tree definition. Features: o Supports playback/capture o Supports 16/20/24 bit PCM o Supports 8k - 96k sample rates o Supports master and slave mode. Signed-off-by: Alison Wang Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..ac4fe4e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,7 @@ +config SND_SOC_FSL_SAI + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + config SND_SOC_FSL_SSI tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8db705b..aaccbee 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o -# Freescale PowerPC SSI/DMA Platform Support +# Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c new file mode 100644 index 0000000..50a797e --- /dev/null +++ b/sound/soc/fsl/fsl_sai.c @@ -0,0 +1,492 @@ +/* + * Freescale ALSA SoC Digital Audio Interface (SAI) driver. + * + * Copyright 2012-2013 Freescale Semiconductor, Inc. + * + * This program is free software, you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation, either version 2 of the License, or(at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "fsl_sai.h" + +static inline u32 sai_readl(struct fsl_sai *sai, + const void __iomem *addr) +{ + u32 val; + + val = __raw_readl(addr); + + if (likely(sai->big_endian_regs)) + val = be32_to_cpu(val); + else + val = le32_to_cpu(val); + rmb(); + + return val; +} + +static inline void sai_writel(struct fsl_sai *sai, + u32 val, void __iomem *addr) +{ + wmb(); + if (likely(sai->big_endian_regs)) + val = cpu_to_be32(val); + else + val = cpu_to_le32(val); + + __raw_writel(val, addr); +} + +static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int fsl_dir) +{ + u32 val_cr2, reg_cr2; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (fsl_dir == FSL_FMT_TRANSMITTER) + reg_cr2 = FSL_SAI_TCR2; + else + reg_cr2 = FSL_SAI_RCR2; + + val_cr2 = sai_readl(sai, sai->base + reg_cr2); + switch (clk_id) { + case FSL_SAI_CLK_BUS: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_BUS; + break; + case FSL_SAI_CLK_MAST1: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1; + break; + case FSL_SAI_CLK_MAST2: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2; + break; + case FSL_SAI_CLK_MAST3: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3; + break; + default: + return -EINVAL; + } + sai_writel(sai, val_cr2, sai->base + reg_cr2); + + return 0; +} + +static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (dir == SND_SOC_CLOCK_IN) + return 0; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); + sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); + sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); + sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); + + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, + FSL_FMT_TRANSMITTER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's transmitter sysclk: %d\n", + ret); + return ret; + } + + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, + FSL_FMT_RECEIVER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's receiver sysclk: %d\n", + ret); + return ret; + } + + clk_disable_unprepare(sai->clk); + + return 0; +} + +static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, + unsigned int fmt, int fsl_dir) +{ + u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (fsl_dir == FSL_FMT_TRANSMITTER) { + reg_cr2 = FSL_SAI_TCR2; + reg_cr3 = FSL_SAI_TCR3; + reg_cr4 = FSL_SAI_TCR4; + } else { + reg_cr2 = FSL_SAI_RCR2; + reg_cr3 = FSL_SAI_RCR3; + reg_cr4 = FSL_SAI_RCR4; + } + + val_cr2 = sai_readl(sai, sai->base + reg_cr2); + val_cr3 = sai_readl(sai, sai->base + reg_cr3); + val_cr4 = sai_readl(sai, sai->base + reg_cr4); + + if (sai->big_endian_data) + val_cr4 |= FSL_SAI_CR4_MF; + else + val_cr4 &= ~FSL_SAI_CR4_MF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val_cr4 |= FSL_SAI_CR4_FSE; + val_cr4 |= FSL_SAI_CR4_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + val_cr4 |= FSL_SAI_CR4_FSP; + val_cr2 &= ~FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_IB_NF: + val_cr4 &= ~FSL_SAI_CR4_FSP; + val_cr2 &= ~FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_NB_IF: + val_cr4 |= FSL_SAI_CR4_FSP; + val_cr2 |= FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_NB_NF: + val_cr4 &= ~FSL_SAI_CR4_FSP; + val_cr2 |= FSL_SAI_CR2_BCP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; + val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; + break; + default: + return -EINVAL; + } + + val_cr3 |= FSL_SAI_CR3_TRCE; + + if (fsl_dir == FSL_FMT_RECEIVER) + val_cr2 |= FSL_SAI_CR2_SYNC; + + sai_writel(sai, val_cr2, sai->base + reg_cr2); + sai_writel(sai, val_cr3, sai->base + reg_cr3); + sai_writel(sai, val_cr4, sai->base + reg_cr4); + + return 0; +} + +static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's transmitter format: %d\n", + ret); + return ret; + } + + ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's receiver format: %d\n", + ret); + return ret; + } + + clk_disable_unprepare(sai->clk); + + return 0; +} + +static int fsl_sai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width; + unsigned int channels = params_channels(params); + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_cr4 = FSL_SAI_TCR4; + reg_cr5 = FSL_SAI_TCR5; + reg_mr = FSL_SAI_TMR; + } else { + reg_cr4 = FSL_SAI_RCR4; + reg_cr5 = FSL_SAI_RCR5; + reg_mr = FSL_SAI_RMR; + } + + val_cr4 = sai_readl(sai, sai->base + reg_cr4); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; + val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; + + val_cr5 = sai_readl(sai, sai->base + reg_cr5); + val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; + val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + word_width = 24; + break; + default: + return -EINVAL; + } + + val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + val_cr5 |= FSL_SAI_CR5_WNW(word_width); + val_cr5 |= FSL_SAI_CR5_W0W(word_width); + + if (sai->big_endian_data) + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); + else + val_cr5 |= FSL_SAI_CR5_FBT(0); + + val_cr4 |= FSL_SAI_CR4_FRSZ(channels); + if (channels == 2 || channels == 1) + val_mr = ~0UL - ((1 << channels) - 1); + else + return -EINVAL; + + sai_writel(sai, val_cr4, sai->base + reg_cr4); + sai_writel(sai, val_cr5, sai->base + reg_cr5); + sai_writel(sai, val_mr, sai->base + reg_mr); + + return 0; +} + +static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + unsigned int tcsr, rcsr; + + tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); + rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tcsr |= FSL_SAI_CSR_FRDE; + rcsr &= ~FSL_SAI_CSR_FRDE; + } else { + rcsr |= FSL_SAI_CSR_FRDE; + tcsr &= ~FSL_SAI_CSR_FRDE; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + tcsr |= FSL_SAI_CSR_TERE; + rcsr |= FSL_SAI_CSR_TERE; + sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!(cpu_dai->playback_active || cpu_dai->capture_active)) { + tcsr &= ~FSL_SAI_CSR_TERE; + rcsr &= ~FSL_SAI_CSR_TERE; + } + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_sai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + ret = clk_prepare_enable(sai->clk); + + return ret; +} + +static void fsl_sai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + clk_disable_unprepare(sai->clk); +} + +static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_sysclk = fsl_sai_set_dai_sysclk, + .set_fmt = fsl_sai_set_dai_fmt, + .hw_params = fsl_sai_hw_params, + .trigger = fsl_sai_trigger, + .startup = fsl_sai_startup, + .shutdown = fsl_sai_shutdown, +}; + +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); + + cpu_dai->playback_dma_data = &sai->dma_params_tx; + cpu_dai->capture_dma_data = &sai->dma_params_rx; + + snd_soc_dai_set_drvdata(cpu_dai, sai); + + return 0; +} + +static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) +{ + cpu_dai->playback_dma_data = NULL; + cpu_dai->capture_dma_data = NULL; + + snd_soc_dai_set_drvdata(cpu_dai, NULL); + + return 0; +} + +static struct snd_soc_dai_driver fsl_sai_dai = { + .probe = fsl_sai_dai_probe, + .remove = fsl_sai_dai_remove, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_SAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_SAI_FORMATS, + }, + .ops = &fsl_sai_pcm_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_component = { + .name = "fsl-sai", +}; + +static int fsl_sai_probe(struct platform_device *pdev) +{ + int ret; + struct fsl_sai *sai; + struct resource *res; + struct device_node *np = pdev->dev.of_node; + + sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); + if (!sai) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + sai->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(sai->base)) + return PTR_ERR(sai->base); + + sai->clk = devm_clk_get(&pdev->dev, "sai"); + if (IS_ERR(sai->clk)) { + dev_err(&pdev->dev, "Cannot get SAI's clock\n"); + return PTR_ERR(sai->clk); + } + + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; + sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; + sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; + sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; + + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); + sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + + platform_set_drvdata(pdev, sai); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + &fsl_sai_dai, 1); + if (ret) + return ret; + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (ret) + return ret; + + return 0; +} + +static int fsl_sai_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + + return 0; +} + +static const struct of_device_id fsl_sai_ids[] = { + { .compatible = "fsl,vf610-sai", }, + { /* sentinel */ } +}; + +static struct platform_driver fsl_sai_driver = { + .probe = fsl_sai_probe, + .remove = fsl_sai_remove, + + .driver = { + .name = "fsl-sai", + .owner = THIS_MODULE, + .of_match_table = fsl_sai_ids, + }, +}; +module_platform_driver(fsl_sai_driver); + +MODULE_DESCRIPTION("Freescale Soc SAI Interface"); +MODULE_AUTHOR("Xiubo Li, "); +MODULE_ALIAS("platform:fsl-sai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h new file mode 100644 index 0000000..41bb62e6 --- /dev/null +++ b/sound/soc/fsl/fsl_sai.h @@ -0,0 +1,114 @@ +/* + * Copyright 2012-2013 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __FSL_SAI_H +#define __FSL_SAI_H + +#include + +#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +/* SAI Transmit/Recieve Control Register */ +#define FSL_SAI_TCSR 0x00 +#define FSL_SAI_RCSR 0x80 +#define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRIE BIT(8) +#define FSL_SAI_CSR_FRDE BIT(0) + +/* SAI Transmit Data/FIFO/MASK Register */ +#define FSL_SAI_TDR 0x20 +#define FSL_SAI_TFR 0x40 +#define FSL_SAI_TMR 0x60 + +/* SAI Recieve Data/FIFO/MASK Register */ +#define FSL_SAI_RDR 0xa0 +#define FSL_SAI_RFR 0xc0 +#define FSL_SAI_RMR 0xe0 + +/* SAI Transmit and Recieve Configuration 1 Register */ +#define FSL_SAI_TCR1 0x04 +#define FSL_SAI_RCR1 0x84 + +/* SAI Transmit and Recieve Configuration 2 Register */ +#define FSL_SAI_TCR2 0x08 +#define FSL_SAI_RCR2 0x88 +#define FSL_SAI_CR2_SYNC BIT(30) +#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_BUS 0 +#define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) +#define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) +#define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_BCP BIT(25) +#define FSL_SAI_CR2_BCD_MSTR BIT(24) + +/* SAI Transmit and Recieve Configuration 3 Register */ +#define FSL_SAI_TCR3 0x0c +#define FSL_SAI_RCR3 0x8c +#define FSL_SAI_CR3_TRCE BIT(16) +#define FSL_SAI_CR3_WDFL(x) (x) +#define FSL_SAI_CR3_WDFL_MASK 0x1f + +/* SAI Transmit and Recieve Configuration 4 Register */ +#define FSL_SAI_TCR4 0x10 +#define FSL_SAI_RCR4 0x90 +#define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) +#define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) +#define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) +#define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_MF BIT(4) +#define FSL_SAI_CR4_FSE BIT(3) +#define FSL_SAI_CR4_FSP BIT(1) +#define FSL_SAI_CR4_FSD_MSTR BIT(0) + +/* SAI Transmit and Recieve Configuration 5 Register */ +#define FSL_SAI_TCR5 0x14 +#define FSL_SAI_RCR5 0x94 +#define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) +#define FSL_SAI_CR5_WNW_MASK (0x1f << 24) +#define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) +#define FSL_SAI_CR5_W0W_MASK (0x1f << 16) +#define FSL_SAI_CR5_FBT(x) ((x) << 8) +#define FSL_SAI_CR5_FBT_MASK (0x1f << 8) + +/* SAI type */ +#define FSL_SAI_DMA BIT(0) +#define FSL_SAI_USE_AC97 BIT(1) +#define FSL_SAI_NET BIT(2) +#define FSL_SAI_TRA_SYN BIT(3) +#define FSL_SAI_REC_SYN BIT(4) +#define FSL_SAI_USE_I2S_SLAVE BIT(5) + +#define FSL_FMT_TRANSMITTER 0 +#define FSL_FMT_RECEIVER 1 + +/* SAI clock sources */ +#define FSL_SAI_CLK_BUS 0 +#define FSL_SAI_CLK_MAST1 1 +#define FSL_SAI_CLK_MAST2 2 +#define FSL_SAI_CLK_MAST3 3 + +/* SAI data transfer numbers per DMA request */ +#define FSL_SAI_MAXBURST_TX 6 +#define FSL_SAI_MAXBURST_RX 6 + +struct fsl_sai { + struct clk *clk; + + void __iomem *base; + + bool big_endian_regs; + bool big_endian_data; + + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; +}; + +#endif /* __FSL_SAI_H */ -- cgit v0.10.2 From b6344859b911990152e5ee411e62b82eb968004f Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 17 Dec 2013 11:24:41 +0800 Subject: ASoC: fsl-sai: Add device tree bindings for Freescale SAI. This adds the Document for Freescale SAI driver under Documentation/devicetree/bindings/sound/. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt new file mode 100644 index 0000000..98611a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -0,0 +1,40 @@ +Freescale Synchronous Audio Interface (SAI). + +The SAI is based on I2S module that used communicating with audio codecs, +which provides a synchronous audio interface that supports fullduplex +serial interfaces with frame synchronization such as I2S, AC97, TDM, and +codec/DSP interfaces. + + +Required properties: +- compatible: Compatible list, contains "fsl,vf610-sai". +- reg: Offset and length of the register set for the device. +- clocks: Must contain an entry for each entry in clock-names. +- clock-names : Must include the "sai" entry. +- dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. +- dma-names : Two dmas have to be defined, "tx" and "rx". +- pinctrl-names: Must contain a "default" entry. +- pinctrl-NNN: One property must exist for each entry in pinctrl-names. + See ../pinctrl/pinctrl-bindings.txt for details of the property values. +- big-endian-regs: If this property is absent, the little endian mode will + be in use as default, or the big endian mode will be in use for all the + device registers. +- big-endian-data: If this property is absent, the little endian mode will + be in use as default, or the big endian mode will be in use for all the + fifo data. + +Example: +sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_SAI2>; + clock-names = "sai"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, + <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; + big-endian-regs; + big-endian-data; +}; -- cgit v0.10.2 From 11db0da831b1e6ae3c1f8743599434281db294db Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 13 Dec 2013 14:43:03 +0800 Subject: ASoC: sgtl5000: clean up sgtl5000_enable_regulators() Function sgtl5000_enable_regulators() is somehow odd in handling the optional external VDDD supply. The driver can only enable this supply on SGTL5000 chip before revision 0x11, and of course when this external VDDD is present. It currently does something like below. 1. Check if regulator_bulk_get() on VDDA, VDDIO and VDDD will fail. If it fails, VDDD must be absent and it falls on internal LDO by calling sgtl5000_replace_vddd_with_ldo(). Otherwise, VDDD is used. And in either case, regulator_bulk_enable() will be called to enable 3 supplies. 2. In case that SGTL5000 revision is later than 0x11, even if external VDDD is present, it has to roll back the 'enable' and 'get' calls with regulator_bulk_disable() and regulator_bulk_free(), and starts over again by calling sgtl5000_replace_vddd_with_ldo() and regulator_bulk_enable(). Such back and forth calls sequence is complicated and unnecessary. Also, since commit 4ddfebd (regulator: core: Provide a dummy regulator with full constraints), regulator_bulk_get() will always succeeds because of the dummy regulator. Thus the VDDD detection is broken. The patch changes the flow to something like the following, which should be more reasonable and clear, and also fix the VDDD detection breakage. 1. Check if we're running a chip before revision 0x11, on which an external VDDD can possibly be an option. 2. If it is an early revision, call regulator_get_optional() to detect whether an external VDDD supply is available. 3. If external VDDD is present, call sgtl5000_replace_vddd_with_ldo() to update sgtl5000->supplies info. 4. Drop regulator_bulk_get() call in sgtl5000_replace_vddd_with_ldo(), and call it in sgtl5000_enable_regulators() no matter it's an external VDDD or internal LDO. 5. Call regulator_bulk_enable() to enable these 3 regulators. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index bd291d2..0fcbe90 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1286,15 +1286,6 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - if (ret) { - ldo_regulator_remove(codec); - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - dev_info(codec->dev, "Using internal LDO instead of VDDD\n"); return 0; } @@ -1305,20 +1296,35 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) int i; int external_vddd = 0; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddd; for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (!ret) - external_vddd = 1; - else { + /* External VDDD only works before revision 0x11 */ + if (sgtl5000->revision < 0x11) { + vddd = regulator_get_optional(codec->dev, "VDDD"); + if (IS_ERR(vddd)) { + /* See if it's just not registered yet */ + if (PTR_ERR(vddd) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + external_vddd = 1; + regulator_put(vddd); + } + } + + if (!external_vddd) { ret = sgtl5000_replace_vddd_with_ldo(codec); if (ret) return ret; } + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_ldo_remove; + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) @@ -1327,37 +1333,13 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - if (external_vddd && sgtl5000->revision >= 0x11) { - /* disable all regulator first */ - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - /* free VDDD regulator */ - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - ret = sgtl5000_replace_vddd_with_ldo(codec); - if (ret) - return ret; - - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) - goto err_regulator_free; - - /* wait for all power rails bring up */ - udelay(10); - } - return 0; err_regulator_free: regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - if (external_vddd) +err_ldo_remove: + if (!external_vddd) ldo_regulator_remove(codec); return ret; -- cgit v0.10.2 From 6f2032a18969d22740a865e0b4f2e48cf5338f36 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:34:45 -0700 Subject: ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device Modify the ep93xx PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index c43fb21..4f900ef 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -63,7 +63,7 @@ static struct snd_soc_ops edb93xx_ops = { static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index efa75b5..cc5583d 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -24,6 +24,8 @@ #include +#include "ep93xx-pcm.h" + /* * Per channel (1-4) registers. */ @@ -394,8 +396,14 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (ret) goto fail; + ret = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (ret) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail: ep93xx_ac97_info = NULL; snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index a57643d..167728a 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -30,6 +30,8 @@ #include #include +#include "ep93xx-pcm.h" + #define EP93XX_I2S_TXCLKCFG 0x00 #define EP93XX_I2S_RXCLKCFG 0x04 #define EP93XX_I2S_GLCTRL 0x0C @@ -405,8 +407,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) if (err) goto fail_put_lrclk; + err = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (err) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index fdb8b8f..198c540 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -23,6 +23,8 @@ #include +#include "ep93xx-pcm.h" + static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -76,27 +78,16 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .prealloc_buffer_size = 131072, }; -static int ep93xx_soc_platform_probe(struct platform_device *pdev) +int devm_ep93xx_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &ep93xx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver ep93xx_pcm_driver = { - .driver = { - .name = "ep93xx-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = ep93xx_soc_platform_probe, -}; - -module_platform_driver(ep93xx_pcm_driver); +EXPORT_SYMBOL_GPL(devm_ep93xx_pcm_platform_register); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h new file mode 100644 index 0000000..b7a12a2 --- /dev/null +++ b/sound/soc/cirrus/ep93xx-pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef __EP93XX_PCM_H__ +#define __EP93XX_PCM_H__ + +int devm_ep93xx_pcm_platform_register(struct device *dev); + +#endif diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c index 4d094d0..822a19a 100644 --- a/sound/soc/cirrus/simone.c +++ b/sound/soc/cirrus/simone.c @@ -27,7 +27,7 @@ static struct snd_soc_dai_link simone_dai = { .cpu_dai_name = "ep93xx-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-ac97", }; static struct snd_soc_card snd_soc_simone = { diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 6904107..29238a7 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, -- cgit v0.10.2 From a8983d4b0a8c439bddae0c9fd1e8a4cf7c402262 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:34:46 -0700 Subject: ASoC: ep93xx: remove custom DMA alloc compat function ep93xx_compat_request_channel() is almost identical to dmaengine_pcm_compat_request_channel(), with the exception that the latter: a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data pointer rather than some custom type. b) dma_data->filter_data rather than dma_data should be passed to snd_dmaengine_pcm_request_channel() as the filter data. Make minor changes to the ep93xx DAI drivers so that those two conditions are met. This allows removal of the custom .compat_request_channel(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index cc5583d..f30dadf 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -19,6 +19,7 @@ #include #include +#include #include #include @@ -97,6 +98,8 @@ struct ep93xx_ac97_info { struct device *dev; void __iomem *regs; struct completion done; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; }; /* currently ALSA only supports a single AC97 device */ @@ -317,8 +320,13 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai) { - dai->playback_dma_data = &ep93xx_ac97_pcm_out; - dai->capture_dma_data = &ep93xx_ac97_pcm_in; + struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); + + info->dma_params_tx.filter_data = &ep93xx_ac97_pcm_out; + info->dma_params_rx.filter_data = &ep93xx_ac97_pcm_in; + + dai->playback_dma_data = &info->dma_params_tx; + dai->capture_dma_data = &info->dma_params_rx; return 0; } diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 167728a..943145f 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -21,6 +21,7 @@ #include #include +#include #include #include #include @@ -63,6 +64,8 @@ struct ep93xx_i2s_info { struct clk *sclk; struct clk *lrclk; void __iomem *regs; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; }; static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { @@ -142,8 +145,15 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) { - dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + + info->dma_params_tx.filter_data = + &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + info->dma_params_rx.filter_data = + &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + dai->playback_dma_data = &info->dma_params_tx; + dai->capture_dma_data = &info->dma_params_rx; return 0; } diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 198c540..ca6698d 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -59,22 +59,9 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } -static struct dma_chan *ep93xx_compat_request_channel( - struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_substream *substream) -{ - struct snd_dmaengine_dai_dma_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter, - dma_data); -} - static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .pcm_hardware = &ep93xx_pcm_hardware, .compat_filter_fn = ep93xx_pcm_dma_filter, - .compat_request_channel = ep93xx_compat_request_channel, .prealloc_buffer_size = 131072, }; -- cgit v0.10.2 From ede38884ac25ed78e43f3480056670963a9980f0 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:35:24 -0700 Subject: ASoC: SPEAr: get rid of spear-pcm-audio struct device Modify the SPEAr PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. This is also required as a pre-cursor to removing spear_pcm_request_chan(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 21a8c95..4627110 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -24,6 +24,7 @@ #include #include #include "spdif_in_regs.h" +#include "spear_pcm.h" struct spdif_in_params { u32 format; @@ -257,8 +258,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } static struct platform_driver spdif_in_driver = { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index b6ef6f7..731a1e0 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -22,6 +22,7 @@ #include #include #include "spdif_out_regs.h" +#include "spear_pcm.h" struct spdif_out_params { u32 rate; @@ -280,6 +281,7 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; + int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -306,8 +308,12 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } #ifdef CONFIG_PM diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9a02141..f288724 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -18,6 +18,7 @@ #include #include #include +#include "spear_pcm.h" static const struct snd_pcm_hardware spear_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -47,26 +48,15 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { .prealloc_buffer_size = 16 * 1024, }; -static int spear_soc_platform_probe(struct platform_device *pdev) +int devm_spear_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &spear_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver spear_pcm_driver = { - .driver = { - .name = "spear-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = spear_soc_platform_probe, -}; - -module_platform_driver(spear_pcm_driver); +EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); MODULE_AUTHOR("Rajeev Kumar "); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:spear-pcm-audio"); diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h new file mode 100644 index 0000000..631e2aa --- /dev/null +++ b/sound/soc/spear/spear_pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef __SPEAR_PCM_H__ +#define __SPEAR_PCM_H__ + +int devm_spear_pcm_platform_register(struct device *dev); + +#endif -- cgit v0.10.2 From e1771bcf99b0dc91f4ba645c1740fd5031702f49 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:35:25 -0700 Subject: ASoC: SPEAr: remove custom DMA alloc compat function spear_pcm_request_chan() is almost identical to dmaengine_pcm_compat_request_channel(), with the exception that the latter: a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data pointer rather than some custom type. b) dma_data->filter_data rather than dma_data should be passed to snd_dmaengine_pcm_request_channel() as the filter data. Make minor changes to the SPEAr DAI drivers so that those two conditions are met. This allows removal of the custom .compat_request_channel(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h index 1b365bf..65aca51 100644 --- a/include/sound/spear_dma.h +++ b/include/sound/spear_dma.h @@ -29,7 +29,6 @@ struct spear_dma_data { dma_addr_t addr; u32 max_burst; enum dma_slave_buswidth addr_width; - bool (*filter)(struct dma_chan *chan, void *slave); }; #endif /* SPEAR_DMA_H */ diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 4627110..4ab442a 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -38,6 +39,8 @@ struct spdif_in_dev { struct device *dev; void (*reset_perip)(void); int irq; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_pcm_config config; }; static void spdif_in_configure(struct spdif_in_dev *host) @@ -54,7 +57,8 @@ static int spdif_in_dai_probe(struct snd_soc_dai *dai) { struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); - dai->capture_dma_data = &host->dma_params; + host->dma_params_rx.filter_data = &host->dma_params; + dai->capture_dma_data = &host->dma_params_rx; return 0; } @@ -245,7 +249,6 @@ static int spdif_in_probe(struct platform_device *pdev) host->dma_params.addr = res_fifo->start; host->dma_params.max_burst = 16; host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - host->dma_params.filter = pdata->filter; host->reset_perip = pdata->reset_perip; host->dev = &pdev->dev; @@ -263,7 +266,8 @@ static int spdif_in_probe(struct platform_device *pdev) if (ret) return ret; - return devm_spear_pcm_platform_register(&pdev->dev); + return devm_spear_pcm_platform_register(&pdev->dev, &host->config, + pdata->filter); } static struct platform_driver spdif_in_driver = { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 731a1e0..fe99f46 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -36,6 +37,8 @@ struct spdif_out_dev { struct spdif_out_params saved_params; u32 running; void __iomem *io_base; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_pcm_config config; }; static void spdif_out_configure(struct spdif_out_dev *host) @@ -245,7 +248,8 @@ static int spdif_soc_dai_probe(struct snd_soc_dai *dai) { struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &host->dma_params; + host->dma_params_tx.filter_data = &host->dma_params; + dai->playback_dma_data = &host->dma_params_tx; return snd_soc_add_dai_controls(dai, spdif_out_controls, ARRAY_SIZE(spdif_out_controls)); @@ -304,7 +308,6 @@ static int spdif_out_probe(struct platform_device *pdev) host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA; host->dma_params.max_burst = 16; host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - host->dma_params.filter = pdata->filter; dev_set_drvdata(&pdev->dev, host); @@ -313,7 +316,8 @@ static int spdif_out_probe(struct platform_device *pdev) if (ret) return ret; - return devm_spear_pcm_platform_register(&pdev->dev); + return devm_spear_pcm_platform_register(&pdev->dev, &host->config, + pdata->filter); } #ifdef CONFIG_PM diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index f288724..0e5a8f3 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -32,26 +32,19 @@ static const struct snd_pcm_hardware spear_pcm_hardware = { .fifo_size = 0, /* fifo size in bytes */ }; -static struct dma_chan *spear_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_substream *substream) -{ - struct spear_dma_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - return snd_dmaengine_pcm_request_channel(dma_data->filter, dma_data); -} - static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { .pcm_hardware = &spear_pcm_hardware, - .compat_request_channel = spear_pcm_request_chan, .prealloc_buffer_size = 16 * 1024, }; -int devm_spear_pcm_platform_register(struct device *dev) +int devm_spear_pcm_platform_register(struct device *dev, + struct snd_dmaengine_pcm_config *config, + bool (*filter)(struct dma_chan *chan, void *slave)) { - return devm_snd_dmaengine_pcm_register(dev, - &spear_dmaengine_pcm_config, + *config = spear_dmaengine_pcm_config; + config->compat_filter_fn = filter; + + return snd_dmaengine_pcm_register(dev, config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h index 631e2aa..9b0ca62 100644 --- a/sound/soc/spear/spear_pcm.h +++ b/sound/soc/spear/spear_pcm.h @@ -17,6 +17,8 @@ #ifndef __SPEAR_PCM_H__ #define __SPEAR_PCM_H__ -int devm_spear_pcm_platform_register(struct device *dev); +int devm_spear_pcm_platform_register(struct device *dev, + struct snd_dmaengine_pcm_config *config, + bool (*filter)(struct dma_chan *chan, void *slave)); #endif -- cgit v0.10.2 From 792b62e705048f4da9a6f570c15093ab669839b6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:11:01 +0000 Subject: mfd: wm5110: Expose DRE control registers Certain use-cases require the DRE to be disabled so expose registers necessary to control the DRE enables. Signed-off-by: Charles Keepax Acked-by: Lee Jones Signed-off-by: Mark Brown diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 338cfbe..abd6713 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -601,6 +601,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x0000043D, 0x0180 }, /* R1085 - DAC Digital Volume 6R */ { 0x0000043E, 0x0080 }, /* R1086 - DAC Volume Limit 6R */ { 0x0000043F, 0x0800 }, /* R1087 - Noise Gate Select 6R */ + { 0x00000440, 0x8FFF }, /* R1088 - DRE Enable */ { 0x00000450, 0x0000 }, /* R1104 - DAC AEC Control 1 */ { 0x00000458, 0x0000 }, /* R1112 - Noise Gate Control */ { 0x00000480, 0x0040 }, /* R1152 - Class W ANC Threshold 1 */ @@ -1631,6 +1632,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_DAC_DIGITAL_VOLUME_6R: case ARIZONA_DAC_VOLUME_LIMIT_6R: case ARIZONA_NOISE_GATE_SELECT_6R: + case ARIZONA_DRE_ENABLE: case ARIZONA_DAC_AEC_CONTROL_1: case ARIZONA_NOISE_GATE_CONTROL: case ARIZONA_PDM_SPK1_CTRL_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 8987814..22916c0 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -3207,6 +3207,10 @@ /* * R1088 (0x440) - DRE Enable */ +#define ARIZONA_DRE3R_ENA 0x0020 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_MASK 0x0020 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_SHIFT 5 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_WIDTH 1 /* DRE3R_ENA */ #define ARIZONA_DRE3L_ENA 0x0010 /* DRE3L_ENA */ #define ARIZONA_DRE3L_ENA_MASK 0x0010 /* DRE3L_ENA */ #define ARIZONA_DRE3L_ENA_SHIFT 4 /* DRE3L_ENA */ -- cgit v0.10.2 From a4c0d2735c66c2363100575324a2f7659fb1f684 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:11:02 +0000 Subject: ASoC: wm5110: Expose switch controls for DRE Certain use-cases require the DRE to be disabled so expose controls for the enables. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ebcbe78..3487ffa 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -371,6 +371,13 @@ SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0), +SOC_DOUBLE("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0), +SOC_DOUBLE("HPOUT3 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- cgit v0.10.2 From fbedc8cbc3c40281cf52ed0e2e5998dea98e2992 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:30:12 +0000 Subject: ASoC: wm5110: Add FSH for ISRCs Currently, the driver only supports configuration of the lower sample rate (FSL) on the ISRCs. With the higher rate being fixed a SYSCLK, this patch adds support for configuring the higher sample rate (FSH). Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6bfd803..56d3ff5 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -498,6 +498,22 @@ const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { EXPORT_SYMBOL_GPL(arizona_rate_val); +const struct soc_enum arizona_isrc_fsh[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1, + ARIZONA_ISRC1_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_1, + ARIZONA_ISRC2_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_1, + ARIZONA_ISRC3_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; +EXPORT_SYMBOL_GPL(arizona_isrc_fsh); + const struct soc_enum arizona_isrc_fsl[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2, ARIZONA_ISRC1_FSL_SHIFT, 0xf, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 6641f3d..99a97c0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -186,6 +186,7 @@ extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; +extern const struct soc_enum arizona_isrc_fsh[]; extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 3487ffa..57be822 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -280,6 +280,9 @@ SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), +SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), +SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), +SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), -- cgit v0.10.2 From 56d37d85438df38e150282baafe52dcd588854c7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:30:13 +0000 Subject: ASoC: wm5110: Add support for ASRC RATE 1 Add support for configuring the sample rate on the SYSCLK side of the ASRC. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 56d3ff5..e4295fe 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -530,6 +530,13 @@ const struct soc_enum arizona_isrc_fsl[] = { }; EXPORT_SYMBOL_GPL(arizona_isrc_fsl); +const struct soc_enum arizona_asrc_rate1 = + SOC_VALUE_ENUM_SINGLE(ARIZONA_ASRC_RATE1, + ARIZONA_ASRC_RATE1_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE - 1, + arizona_rate_text, arizona_rate_val); +EXPORT_SYMBOL_GPL(arizona_asrc_rate1); + static const char *arizona_vol_ramp_text[] = { "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", "15ms/6dB", "30ms/6dB", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 99a97c0..10b3984 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -187,6 +187,7 @@ extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; extern const struct soc_enum arizona_isrc_fsh[]; +extern const struct soc_enum arizona_asrc_rate1; extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 57be822..f3d96ea 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -283,6 +283,7 @@ SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), +SOC_VALUE_ENUM("ASRC RATE 1", arizona_asrc_rate1), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), -- cgit v0.10.2 From e1acb40a3addc9aceb4600f04c9c86b50770b9b8 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 19 Dec 2013 11:59:54 +0800 Subject: ASoC: simple-card: Use devm_snd_soc_register_card() Makes the code slightly shorter. Signed-off-by: Xiubo Li Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7a9b6b4..3d190d0 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -234,14 +234,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.num_links = 1; cinfo->snd_card.dev = &pdev->dev; - return snd_soc_register_card(&cinfo->snd_card); -} - -static int asoc_simple_card_remove(struct platform_device *pdev) -{ - struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; - - return snd_soc_unregister_card(&cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { @@ -257,7 +250,6 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, - .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); -- cgit v0.10.2 From 05004cb4cd06127bb8ff70d5ab5a915103828e9d Mon Sep 17 00:00:00 2001 From: Matthew Garrett Date: Wed, 18 Dec 2013 13:50:10 +0000 Subject: ASoC: fsl/mxs: Remove unnecessarily gendered language The kernel as a number of cases of gendered language. The majority of these refer to objects that don't have gender in English, and so I've replaced them with "it" and "its". Some refer to people (developers or users), and I've replaced these with the singular "they" variant. Some are simply typos that I've fixed up. I've left cases where gendered language was used to refer to specific individuals, was a quote or is part of license text. Signed-off-by: Matthew Garrett Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index fb9bb9e..d570f8c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -852,7 +852,7 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) } /** - * find_ssi_node -- returns the SSI node that points to his DMA channel node + * find_ssi_node -- returns the SSI node that points to its DMA channel node * * Although this DMA driver attempts to operate independently of the other * devices, it still needs to determine some information about the SSI device diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54e622a..d6cb9a5 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -50,9 +50,9 @@ static struct mxs_saif *mxs_saif[2]; * This also means that both SAIFs must operate at the same sample rate. * * We abstract this as each saif has a master, the master could be - * himself or other saifs. In the generic saif driver, saif does not need - * to know the different clkmux. Saif only needs to know who is his master - * and operating his master to generate the proper clock rate for him. + * itself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is its master + * and operating its master to generate the proper clock rate for it. * The master id is provided in mach-specific layer according to different * clkmux setting. */ @@ -76,7 +76,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK * is provided by other SAIF, we provide a interface here to get its master * from its master_id. - * Note that the master could be himself. + * Note that the master could be itself. */ static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) { @@ -516,7 +516,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } /* - * If the saif's master is not himself, we also need to enable + * If the saif's master is not itself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { -- cgit v0.10.2 From aafa85e71a75fdea9076c5e0f7fe09e12568c9a4 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 12 Dec 2013 18:44:45 +0800 Subject: ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569..b2ebaf8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include #include #include +#include #include #include #include @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -502,6 +511,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } /** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + +/** * fsl_ssi_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69..e6b6324 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- cgit v0.10.2 From c7f9129d22940720141d1f1e958a51142eff9d21 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:45 +0200 Subject: mfd: twl6040: reg_defaults support for regmap Add reg_defaults to regmap and at the same time implement proper power state handling with using regcache_cache_only(), regcache_sync() and regcache_mark_dirty(). This will make sure that we do not need to do restore operations in child drivers anymore. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Lee Jones diff --git a/drivers/mfd/twl6040.c b/drivers/mfd/twl6040.c index 0779d5a..51b6df1 100644 --- a/drivers/mfd/twl6040.c +++ b/drivers/mfd/twl6040.c @@ -44,6 +44,54 @@ #define VIBRACTRL_MEMBER(reg) ((reg == TWL6040_REG_VIBCTLL) ? 0 : 1) #define TWL6040_NUM_SUPPLIES (2) +static struct reg_default twl6040_defaults[] = { + { 0x01, 0x4B }, /* REG_ASICID (ro) */ + { 0x02, 0x00 }, /* REG_ASICREV (ro) */ + { 0x03, 0x00 }, /* REG_INTID */ + { 0x04, 0x00 }, /* REG_INTMR */ + { 0x05, 0x00 }, /* REG_NCPCTRL */ + { 0x06, 0x00 }, /* REG_LDOCTL */ + { 0x07, 0x60 }, /* REG_HPPLLCTL */ + { 0x08, 0x00 }, /* REG_LPPLLCTL */ + { 0x09, 0x4A }, /* REG_LPPLLDIV */ + { 0x0A, 0x00 }, /* REG_AMICBCTL */ + { 0x0B, 0x00 }, /* REG_DMICBCTL */ + { 0x0C, 0x00 }, /* REG_MICLCTL */ + { 0x0D, 0x00 }, /* REG_MICRCTL */ + { 0x0E, 0x00 }, /* REG_MICGAIN */ + { 0x0F, 0x1B }, /* REG_LINEGAIN */ + { 0x10, 0x00 }, /* REG_HSLCTL */ + { 0x11, 0x00 }, /* REG_HSRCTL */ + { 0x12, 0x00 }, /* REG_HSGAIN */ + { 0x13, 0x00 }, /* REG_EARCTL */ + { 0x14, 0x00 }, /* REG_HFLCTL */ + { 0x15, 0x00 }, /* REG_HFLGAIN */ + { 0x16, 0x00 }, /* REG_HFRCTL */ + { 0x17, 0x00 }, /* REG_HFRGAIN */ + { 0x18, 0x00 }, /* REG_VIBCTLL */ + { 0x19, 0x00 }, /* REG_VIBDATL */ + { 0x1A, 0x00 }, /* REG_VIBCTLR */ + { 0x1B, 0x00 }, /* REG_VIBDATR */ + { 0x1C, 0x00 }, /* REG_HKCTL1 */ + { 0x1D, 0x00 }, /* REG_HKCTL2 */ + { 0x1E, 0x00 }, /* REG_GPOCTL */ + { 0x1F, 0x00 }, /* REG_ALB */ + { 0x20, 0x00 }, /* REG_DLB */ + /* 0x28, REG_TRIM1 */ + /* 0x29, REG_TRIM2 */ + /* 0x2A, REG_TRIM3 */ + /* 0x2B, REG_HSOTRIM */ + /* 0x2C, REG_HFOTRIM */ + { 0x2D, 0x08 }, /* REG_ACCCTL */ + { 0x2E, 0x00 }, /* REG_STATUS (ro) */ +}; + +struct reg_default twl6040_patch[] = { + /* Select I2C bus access to dual access registers */ + { TWL6040_REG_ACCCTL, 0x09 }, +}; + + static bool twl6040_has_vibra(struct device_node *node) { #ifdef CONFIG_OF @@ -238,6 +286,9 @@ int twl6040_power(struct twl6040 *twl6040, int on) if (twl6040->power_count++) goto out; + /* Allow writes to the chip */ + regcache_cache_only(twl6040->regmap, false); + if (gpio_is_valid(twl6040->audpwron)) { /* use automatic power-up sequence */ ret = twl6040_power_up_automatic(twl6040); @@ -253,6 +304,10 @@ int twl6040_power(struct twl6040 *twl6040, int on) goto out; } } + + /* Sync with the HW */ + regcache_sync(twl6040->regmap); + /* Default PLL configuration after power up */ twl6040->pll = TWL6040_SYSCLK_SEL_LPPLL; twl6040->sysclk = 19200000; @@ -279,6 +334,11 @@ int twl6040_power(struct twl6040 *twl6040, int on) /* use manual power-down sequence */ twl6040_power_down_manual(twl6040); } + + /* Set regmap to cache only and mark it as dirty */ + regcache_cache_only(twl6040->regmap, true); + regcache_mark_dirty(twl6040->regmap); + twl6040->sysclk = 0; twl6040->mclk = 0; } @@ -490,9 +550,24 @@ static bool twl6040_readable_reg(struct device *dev, unsigned int reg) static bool twl6040_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case TWL6040_REG_VIBCTLL: - case TWL6040_REG_VIBCTLR: - case TWL6040_REG_INTMR: + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_INTID: + case TWL6040_REG_LPPLLCTL: + case TWL6040_REG_HPPLLCTL: + case TWL6040_REG_STATUS: + return true; + default: + return false; + } +} + +static bool twl6040_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: return false; default: return true; @@ -502,10 +577,15 @@ static bool twl6040_volatile_reg(struct device *dev, unsigned int reg) static struct regmap_config twl6040_regmap_config = { .reg_bits = 8, .val_bits = 8, + + .reg_defaults = twl6040_defaults, + .num_reg_defaults = ARRAY_SIZE(twl6040_defaults), + .max_register = TWL6040_REG_STATUS, /* 0x2e */ .readable_reg = twl6040_readable_reg, .volatile_reg = twl6040_volatile_reg, + .writeable_reg = twl6040_writeable_reg, .cache_type = REGCACHE_RBTREE, }; @@ -624,6 +704,8 @@ static int twl6040_probe(struct i2c_client *client, /* dual-access registers controlled by I2C only */ twl6040_set_bits(twl6040, TWL6040_REG_ACCCTL, TWL6040_I2CSEL); + regmap_register_patch(twl6040->regmap, twl6040_patch, + ARRAY_SIZE(twl6040_patch)); /* * The main functionality of twl6040 to provide audio on OMAP4+ systems. @@ -656,6 +738,10 @@ static int twl6040_probe(struct i2c_client *client, cell->name = "twl6040-gpo"; children++; + /* The chip is powered down so mark regmap to cache only and dirty */ + regcache_cache_only(twl6040->regmap, true); + regcache_mark_dirty(twl6040->regmap); + ret = mfd_add_devices(&client->dev, -1, twl6040->cells, children, NULL, 0, NULL); if (ret) -- cgit v0.10.2 From 0d35d080ac93f317b6c47180d75c8e1a8109b4c4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:43 +0200 Subject: ASoC: twl6040: Rename twl6040_is_path_unmuted -> twl6040_can_write_to_chip Matches more precisely of the functionality. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index f2f4bcb..ef13a50 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -219,8 +219,8 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, return value; } -static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, - unsigned int reg) +static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, + unsigned int reg) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); @@ -250,7 +250,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (twl6040_is_path_unmuted(codec, reg)) + if (twl6040_can_write_to_chip(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; -- cgit v0.10.2 From 53509108f7372f786576d7d43f8f881cdf51d38a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:44 +0200 Subject: ASoC: twl6040: Custom caching for sensitive DL1/2 path registers Introduce a small register cache for registers which needs special caching to reduce pop noise: TWL6040_REG_HSLCTL, TWL6040_REG_HSRCTL, TWL6040_REG_EARCTL, TWL6040_REG_HFLCTL and TWL6040_REG_HFRCTL. Switch over and use the new small cache for these registers instead of the main reg_cache. This is in preparation to remove the local ASoC reg_cache from the driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index ef13a50..fb8c65b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -72,6 +72,7 @@ struct twl6040_data { int hs_power_mode_locked; bool dl1_unmuted; bool dl2_unmuted; + u8 dl12_cache[TWL6040_REG_HFRCTL - TWL6040_REG_HSLCTL + 1]; unsigned int clk_in; unsigned int sysclk; struct twl6040_jack_data hs_jack; @@ -174,18 +175,62 @@ static struct snd_pcm_hw_constraint_list sysclk_constraints[] = { { .count = ARRAY_SIZE(hp_rates), .list = hp_rates, }, }; +static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec, + u8 reg, u8 *value) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (reg) { + case TWL6040_REG_HSLCTL: + case TWL6040_REG_HSRCTL: + case TWL6040_REG_EARCTL: + case TWL6040_REG_HFLCTL: + case TWL6040_REG_HFRCTL: + *value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL]; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + /* * read twl6040 register cache */ static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { u8 *cache = codec->reg_cache; + u8 value; if (reg >= TWL6040_CACHEREGNUM) return -EIO; - return cache[reg]; + if (twl6040_read_dl12_cache(codec, reg, &value)) + value = cache[reg]; + + return value; +} + +static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + + switch (reg) { + case TWL6040_REG_HSLCTL: + case TWL6040_REG_HSRCTL: + case TWL6040_REG_EARCTL: + case TWL6040_REG_HFLCTL: + case TWL6040_REG_HFRCTL: + priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value; + break; + default: + break; + } } /* @@ -199,6 +244,8 @@ static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return; cache[reg] = value; + + twl6040_update_dl12_cache(codec, reg, value); } /* -- cgit v0.10.2 From 79ae5130381cb7d3e74c162123392ba6067c218e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:46 +0200 Subject: ASoC: twl6040: Remove register restore functionality The MFD core takes care of the restore via standard regmap API, no need to do this anymore here. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index fb8c65b..b07839b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -133,22 +133,6 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* REG_STATUS 0x2E (ro) */ }; -/* List of registers to be restored after power up */ -static const int twl6040_restore_list[] = { - TWL6040_REG_MICLCTL, - TWL6040_REG_MICRCTL, - TWL6040_REG_MICGAIN, - TWL6040_REG_LINEGAIN, - TWL6040_REG_HSLCTL, - TWL6040_REG_HSRCTL, - TWL6040_REG_HSGAIN, - TWL6040_REG_EARCTL, - TWL6040_REG_HFLCTL, - TWL6040_REG_HFLGAIN, - TWL6040_REG_HFRCTL, - TWL6040_REG_HFRGAIN, -}; - /* set of rates for each pll: low-power and high-performance */ static unsigned int lp_rates[] = { 8000, @@ -335,17 +319,6 @@ static void twl6040_init_chip(struct snd_soc_codec *codec) twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0); } -static void twl6040_restore_regs(struct snd_soc_codec *codec) -{ - u8 *cache = codec->reg_cache; - int reg, i; - - for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) { - reg = twl6040_restore_list[i]; - twl6040_write(codec, reg, cache[reg]); - } -} - /* set headset dac and driver power mode */ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { @@ -978,8 +951,6 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 1; - twl6040_restore_regs(codec); - /* Set external boost GPO */ twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02); break; -- cgit v0.10.2 From 626bcacb89f93b2145f3a705a342067a77347a99 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:47 +0200 Subject: ASoC: twl6040: Remove self managed local reg_cache support We can rely on mfd driver to manage the register caching via regmap. The driver still need to cache some registers associated with DL1/2 routes. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b07839b..0afe8be 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -80,59 +80,6 @@ struct twl6040_data { struct mutex mutex; }; -/* - * twl6040 register cache & default register settings - */ -static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { - 0x00, /* not used 0x00 */ - 0x4B, /* REG_ASICID 0x01 (ro) */ - 0x00, /* REG_ASICREV 0x02 (ro) */ - 0x00, /* REG_INTID 0x03 */ - 0x00, /* REG_INTMR 0x04 */ - 0x00, /* REG_NCPCTRL 0x05 */ - 0x00, /* REG_LDOCTL 0x06 */ - 0x60, /* REG_HPPLLCTL 0x07 */ - 0x00, /* REG_LPPLLCTL 0x08 */ - 0x4A, /* REG_LPPLLDIV 0x09 */ - 0x00, /* REG_AMICBCTL 0x0A */ - 0x00, /* REG_DMICBCTL 0x0B */ - 0x00, /* REG_MICLCTL 0x0C */ - 0x00, /* REG_MICRCTL 0x0D */ - 0x00, /* REG_MICGAIN 0x0E */ - 0x1B, /* REG_LINEGAIN 0x0F */ - 0x00, /* REG_HSLCTL 0x10 */ - 0x00, /* REG_HSRCTL 0x11 */ - 0x00, /* REG_HSGAIN 0x12 */ - 0x00, /* REG_EARCTL 0x13 */ - 0x00, /* REG_HFLCTL 0x14 */ - 0x00, /* REG_HFLGAIN 0x15 */ - 0x00, /* REG_HFRCTL 0x16 */ - 0x00, /* REG_HFRGAIN 0x17 */ - 0x00, /* REG_VIBCTLL 0x18 */ - 0x00, /* REG_VIBDATL 0x19 */ - 0x00, /* REG_VIBCTLR 0x1A */ - 0x00, /* REG_VIBDATR 0x1B */ - 0x00, /* REG_HKCTL1 0x1C */ - 0x00, /* REG_HKCTL2 0x1D */ - 0x00, /* REG_GPOCTL 0x1E */ - 0x00, /* REG_ALB 0x1F */ - 0x00, /* REG_DLB 0x20 */ - 0x00, /* not used 0x21 */ - 0x00, /* not used 0x22 */ - 0x00, /* not used 0x23 */ - 0x00, /* not used 0x24 */ - 0x00, /* not used 0x25 */ - 0x00, /* not used 0x26 */ - 0x00, /* not used 0x27 */ - 0x00, /* REG_TRIM1 0x28 */ - 0x00, /* REG_TRIM2 0x29 */ - 0x00, /* REG_TRIM3 0x2A */ - 0x00, /* REG_HSOTRIM 0x2B */ - 0x00, /* REG_HFOTRIM 0x2C */ - 0x09, /* REG_ACCCTL 0x2D */ - 0x00, /* REG_STATUS 0x2E (ro) */ -}; - /* set of rates for each pll: low-power and high-performance */ static unsigned int lp_rates[] = { 8000, @@ -159,11 +106,14 @@ static struct snd_pcm_hw_constraint_list sysclk_constraints[] = { { .count = ARRAY_SIZE(hp_rates), .list = hp_rates, }, }; -static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec, - u8 reg, u8 *value) +static unsigned int twl6040_read(struct snd_soc_codec *codec, unsigned int reg) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret = 0; + struct twl6040 *twl6040 = codec->control_data; + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; switch (reg) { case TWL6040_REG_HSLCTL: @@ -171,36 +121,18 @@ static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec, case TWL6040_REG_EARCTL: case TWL6040_REG_HFLCTL: case TWL6040_REG_HFRCTL: - *value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL]; + value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL]; break; default: - ret = -EINVAL; + value = twl6040_reg_read(twl6040, reg); break; } - return ret; -} - -/* - * read twl6040 register cache - */ -static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 *cache = codec->reg_cache; - u8 value; - - if (reg >= TWL6040_CACHEREGNUM) - return -EIO; - - if (twl6040_read_dl12_cache(codec, reg, &value)) - value = cache[reg]; - return value; } -static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, - u8 reg, u8 value) +static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, + unsigned int reg) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); @@ -208,50 +140,18 @@ static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, case TWL6040_REG_HSLCTL: case TWL6040_REG_HSRCTL: case TWL6040_REG_EARCTL: + /* DL1 path */ + return priv->dl1_unmuted; case TWL6040_REG_HFLCTL: case TWL6040_REG_HFRCTL: - priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value; - break; + return priv->dl2_unmuted; default: - break; + return 1; } } -/* - * write twl6040 register cache - */ -static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u8 value) -{ - u8 *cache = codec->reg_cache; - - if (reg >= TWL6040_CACHEREGNUM) - return; - cache[reg] = value; - - twl6040_update_dl12_cache(codec, reg, value); -} - -/* - * read from twl6040 hardware register - */ -static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct twl6040 *twl6040 = codec->control_data; - u8 value; - - if (reg >= TWL6040_CACHEREGNUM) - return -EIO; - - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); - - return value; -} - -static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, - unsigned int reg) +static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); @@ -259,19 +159,15 @@ static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, case TWL6040_REG_HSLCTL: case TWL6040_REG_HSRCTL: case TWL6040_REG_EARCTL: - /* DL1 path */ - return priv->dl1_unmuted; case TWL6040_REG_HFLCTL: case TWL6040_REG_HFRCTL: - return priv->dl2_unmuted; + priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value; + break; default: - return 1; + break; } } -/* - * write to the twl6040 register space - */ static int twl6040_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -280,7 +176,7 @@ static int twl6040_write(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - twl6040_write_reg_cache(codec, reg, value); + twl6040_update_dl12_cache(codec, reg, value); if (twl6040_can_write_to_chip(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else @@ -289,34 +185,27 @@ static int twl6040_write(struct snd_soc_codec *codec, static void twl6040_init_chip(struct snd_soc_codec *codec) { - struct twl6040 *twl6040 = codec->control_data; - u8 val; - - /* Update reg_cache: ASICREV, and TRIM values */ - val = twl6040_get_revid(twl6040); - twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); - - twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM1); - twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM2); - twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM3); - twl6040_read_reg_volatile(codec, TWL6040_REG_HSOTRIM); - twl6040_read_reg_volatile(codec, TWL6040_REG_HFOTRIM); + twl6040_read(codec, TWL6040_REG_TRIM1); + twl6040_read(codec, TWL6040_REG_TRIM2); + twl6040_read(codec, TWL6040_REG_TRIM3); + twl6040_read(codec, TWL6040_REG_HSOTRIM); + twl6040_read(codec, TWL6040_REG_HFOTRIM); /* Change chip defaults */ /* No imput selected for microphone amplifiers */ - twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); - twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18); + twl6040_write(codec, TWL6040_REG_MICLCTL, 0x18); + twl6040_write(codec, TWL6040_REG_MICRCTL, 0x18); /* * We need to lower the default gain values, so the ramp code * can work correctly for the first playback. * This reduces the pop noise heard at the first playback. */ - twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff); - twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e); - twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d); - twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d); - twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0); + twl6040_write(codec, TWL6040_REG_HSGAIN, 0xff); + twl6040_write(codec, TWL6040_REG_EARCTL, 0x1e); + twl6040_write(codec, TWL6040_REG_HFLGAIN, 0x1d); + twl6040_write(codec, TWL6040_REG_HFRGAIN, 0x1d); + twl6040_write(codec, TWL6040_REG_LINEGAIN, 0); } /* set headset dac and driver power mode */ @@ -325,8 +214,8 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) int hslctl, hsrctl; int mask = TWL6040_HSDRVMODE | TWL6040_HSDACMODE; - hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); - hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL); if (high_perf) { hslctl &= ~mask; @@ -353,8 +242,8 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, * Both HS DAC need to be turned on (before the HS driver) and off at * the same time. */ - hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); - hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL); if (SND_SOC_DAPM_EVENT_ON(event)) { hslctl |= TWL6040_HSDACENA; hsrctl |= TWL6040_HSDACENA; @@ -399,7 +288,7 @@ static void twl6040_hs_jack_report(struct snd_soc_codec *codec, mutex_lock(&priv->mutex); /* Sync status */ - status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS); + status = twl6040_read(codec, TWL6040_REG_STATUS); if (status & TWL6040_PLUGCOMP) snd_soc_jack_report(jack, report, report); else @@ -451,7 +340,7 @@ static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, unsigned int val; /* Do not allow changes while Input/FF efect is running */ - val = twl6040_read_reg_volatile(codec, e->reg); + val = twl6040_read(codec, e->reg); if (val & TWL6040_VIBENA && !(val & TWL6040_VIBSEL)) return -EBUSY; @@ -676,7 +565,7 @@ int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim) if (unlikely(trim >= TWL6040_TRIM_INVAL)) return -EINVAL; - return twl6040_read_reg_cache(codec, TWL6040_REG_TRIM1 + trim); + return twl6040_read(codec, TWL6040_REG_TRIM1 + trim); } EXPORT_SYMBOL_GPL(twl6040_get_trim_value); @@ -1071,9 +960,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i switch (id) { case TWL6040_DAI_DL1: - hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); - hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); - earctl = twl6040_read_reg_cache(codec, TWL6040_REG_EARCTL); + hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL); + earctl = twl6040_read(codec, TWL6040_REG_EARCTL); if (mute) { /* Power down drivers and DACs */ @@ -1089,8 +978,8 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i priv->dl1_unmuted = !mute; break; case TWL6040_DAI_DL2: - hflctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFLCTL); - hfrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFRCTL); + hflctl = twl6040_read(codec, TWL6040_REG_HFLCTL); + hfrctl = twl6040_read(codec, TWL6040_REG_HFRCTL); if (mute) { /* Power down drivers and DACs */ @@ -1227,6 +1116,7 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; + struct twl6040 *twl6040 = dev_get_drvdata(codec->dev->parent); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; @@ -1238,7 +1128,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, priv); priv->codec = codec; - codec->control_data = dev_get_drvdata(codec->dev->parent); + codec->control_data = twl6040; priv->plug_irq = platform_get_irq(pdev, 0); if (priv->plug_irq < 0) { @@ -1258,10 +1148,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) return ret; } + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); twl6040_init_chip(codec); - /* power on device */ - return twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; } static int twl6040_remove(struct snd_soc_codec *codec) @@ -1279,12 +1169,9 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .remove = twl6040_remove, .suspend = twl6040_suspend, .resume = twl6040_resume, - .read = twl6040_read_reg_cache, + .read = twl6040_read, .write = twl6040_write, .set_bias_level = twl6040_set_bias_level, - .reg_cache_size = ARRAY_SIZE(twl6040_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = twl6040_reg, .ignore_pmdown_time = true, .controls = twl6040_snd_controls, -- cgit v0.10.2 From f467a0f513ad81998f0cad1022684a273d5743f7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:08 +0100 Subject: ASoC: au1x: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Tested-by: Manuel Lauss Signed-off-by: Mark Brown diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 3b4eafa..17a24d8 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -65,19 +65,10 @@ struct au1xpsc_audio_dmadata { #define AU1XPSC_PERIOD_MIN_BYTES 1024 #define AU1XPSC_BUFFER_MIN_BYTES 65536 -#define AU1XPSC_PCM_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ - SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ - SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ - 0) - /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, - .formats = AU1XPSC_PCM_FMTS, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, .periods_min = 2, diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index befd107..e920b60 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -21,14 +21,6 @@ #include "psc.h" -#define ALCHEMY_PCM_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ - SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ - SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ - 0) - struct pcm_period { u32 start; u32 relative_end; /* relative to start of buffer */ @@ -171,12 +163,6 @@ static irqreturn_t au1000_dma_interrupt(int irq, void *ptr) static const struct snd_pcm_hardware alchemy_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, - .formats = ALCHEMY_PCM_FMTS, - .rates = SNDRV_PCM_RATE_8000_192000, - .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_192000, - .channels_min = 2, - .channels_max = 2, .period_bytes_min = 1024, .period_bytes_max = 16 * 1024 - 1, .periods_min = 4, -- cgit v0.10.2 From 5e8154332f48f92f37824577c88e400b5e0cd56d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:22 +0100 Subject: ASoC: tegra: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Tested-by: Stephen Warren Acked-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 7b2d23b..c09ffd1 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -42,9 +42,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .channels_min = 2, - .channels_max = 2, .period_bytes_min = 1024, .period_bytes_max = PAGE_SIZE, .periods_min = 2, -- cgit v0.10.2 From a6af47ae5399baf4f5a2426b2121c1bcb9da4019 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 12:17:38 +0800 Subject: ASoC: fsl-sai: Remove fsl_sai_remove() There is no need of this function and makes the code slightly shorter Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e..1868ec3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -385,19 +385,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) -{ - cpu_dai->playback_dma_data = NULL; - cpu_dai->capture_dma_data = NULL; - - snd_soc_dai_set_drvdata(cpu_dai, NULL); - - return 0; -} - static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, - .remove = fsl_sai_dai_remove, .playback = { .channels_min = 1, .channels_max = 2, -- cgit v0.10.2 From e5180df3960b6130f17f3c5ab50d23674cdb2b5a Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 12:30:26 +0800 Subject: ASoC: fsl-sai: Use devm_snd_dmaengine_pcm_register() Makes the code slightly shorter Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1868ec3..262d310 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -443,19 +443,8 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); - if (ret) - return ret; - - return 0; -} - -static int fsl_sai_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - - return 0; } static const struct of_device_id fsl_sai_ids[] = { @@ -465,8 +454,6 @@ static const struct of_device_id fsl_sai_ids[] = { static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, - .remove = fsl_sai_remove, - .driver = { .name = "fsl-sai", .owner = THIS_MODULE, -- cgit v0.10.2 From dd9f40602e96353c210805a99abd9af6abd28473 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 12:35:33 +0800 Subject: ASoC: fsl-sai: Use snd_soc_dai_init_dma_data() Makes the code slightly shorter Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 262d310..b8cdbf8 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -377,8 +377,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - cpu_dai->playback_dma_data = &sai->dma_params_tx; - cpu_dai->capture_dma_data = &sai->dma_params_rx; + snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, + &sai->dma_params_rx); snd_soc_dai_set_drvdata(cpu_dai, sai); -- cgit v0.10.2 From 1fb2d9d7465bcbb519c582fa4a3bd04ff4fce2d2 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:00 +0800 Subject: ASoC: fsl_sai: Keep symmetry for clk_enable() and clk_disable() There are two functions haven't clk_disable_unprepare() if having error. Thus fix them. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b8cdbf8..69a375f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -111,7 +111,7 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, dev_err(cpu_dai->dev, "Cannot set SAI's transmitter sysclk: %d\n", ret); - return ret; + goto err_clk; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, @@ -120,12 +120,13 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, dev_err(cpu_dai->dev, "Cannot set SAI's receiver sysclk: %d\n", ret); - return ret; + goto err_clk; } +err_clk: clk_disable_unprepare(sai->clk); - return 0; + return ret; } static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, @@ -222,7 +223,7 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) dev_err(cpu_dai->dev, "Cannot set SAI's transmitter format: %d\n", ret); - return ret; + goto err_clk; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); @@ -230,12 +231,13 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) dev_err(cpu_dai->dev, "Cannot set SAI's receiver format: %d\n", ret); - return ret; + goto err_clk; } +err_clk: clk_disable_unprepare(sai->clk); - return 0; + return ret; } static int fsl_sai_hw_params(struct snd_pcm_substream *substream, -- cgit v0.10.2 From 1d7003092771bd2feec30e2f3e5a06aa33479e08 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:01 +0800 Subject: ASoC: fsl_sai: Use snd_pcm_format_width() Use common helper function snd_pcm_format_width() to make code neater. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 69a375f..e68102e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -244,9 +244,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width; + u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr; unsigned int channels = params_channels(params); struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 word_width = snd_pcm_format_width(params_format(params)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg_cr4 = FSL_SAI_TCR4; @@ -267,20 +268,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - word_width = 16; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - word_width = 20; - break; - case SNDRV_PCM_FORMAT_S24_LE: - word_width = 24; - break; - default: - return -EINVAL; - } - val_cr4 |= FSL_SAI_CR4_SYWD(word_width); val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); -- cgit v0.10.2 From d22e28cce80a93578787d273bf1fa26a2be2636b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:02 +0800 Subject: ASoC: fsl_sai: Drop useless channels check in hw_params() SAi only supports two data channels on hardware level and the driver also does register the min->1 and max->2, so no need to check channels. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index e68102e..8450bff 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -278,10 +278,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_FBT(0); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); - if (channels == 2 || channels == 1) - val_mr = ~0UL - ((1 << channels) - 1); - else - return -EINVAL; + val_mr = ~0UL - ((1 << channels) - 1); sai_writel(sai, val_cr4, sai->base + reg_cr4); sai_writel(sai, val_cr5, sai->base + reg_cr5); -- cgit v0.10.2 From 15b29dae6604d2d2daf586429ff12f26272a868a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:03 +0800 Subject: ASoC: fsl_sai: Drop useless ret in startup() We can save this ret to make the code neater. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8450bff..fc4cd95 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -334,12 +334,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - int ret; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - ret = clk_prepare_enable(sai->clk); - - return ret; + return clk_prepare_enable(sai->clk); } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, -- cgit v0.10.2 From 190af12dad975f2ea7d69d1c5c9d36fec64da767 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:04 +0800 Subject: ASoC: fsl_sai: Make dev_err information neater Since using dev_err() there's no need to mention SAI any more, it will print the full name of the driver -- fsl_sai. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fc4cd95..68d666b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -108,18 +108,14 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's transmitter sysclk: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); goto err_clk; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_RECEIVER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's receiver sysclk: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret); goto err_clk; } @@ -220,17 +216,13 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's transmitter format: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret); goto err_clk; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's receiver format: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret); goto err_clk; } -- cgit v0.10.2 From 4e3a99f5b004b30bc604d82e5498700649148e0d Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:05 +0800 Subject: ASoC: fsl_sai: Sort local variable in general way Generally we would write code for local variable like: static new_func() { struct xxx *yyy; ... int ret; } But this driver only follows this pattern for some functions, not all. Thus this patch sorts the local variable in the general way. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 68d666b..b72132f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -53,8 +53,8 @@ static inline void sai_writel(struct fsl_sai *sai, static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { - u32 val_cr2, reg_cr2; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 val_cr2, reg_cr2; if (fsl_dir == FSL_FMT_TRANSMITTER) reg_cr2 = FSL_SAI_TCR2; @@ -90,8 +90,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - int ret; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + int ret; if (dir == SND_SOC_CLOCK_IN) return 0; @@ -128,8 +128,8 @@ err_clk: static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { - u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; if (fsl_dir == FSL_FMT_TRANSMITTER) { reg_cr2 = FSL_SAI_TCR2; @@ -207,8 +207,8 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - int ret; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + int ret; ret = clk_prepare_enable(sai->clk); if (ret) @@ -236,9 +236,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr; unsigned int channels = params_channels(params); - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); u32 word_width = snd_pcm_format_width(params_format(params)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -383,10 +383,10 @@ static const struct snd_soc_component_driver fsl_component = { static int fsl_sai_probe(struct platform_device *pdev) { - int ret; + struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; struct resource *res; - struct device_node *np = pdev->dev.of_node; + int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) -- cgit v0.10.2 From d754fa9ad18d16209c276fc6241aa2d10f819ede Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:09 +0100 Subject: ASoC: blackfin: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 1d4c676..cdb8ee7 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -107,7 +107,6 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { #endif SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 0x10000, .periods_min = 1, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 2a5b434..a3881c4 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -52,9 +52,6 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 0x10000, .periods_min = 1, -- cgit v0.10.2 From f52c91921553be26c7a0de13daa0d18ef46655ff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:10 +0100 Subject: ASoC: davinci: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index fb5d107..14145cd 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -46,33 +46,11 @@ static void print_buf_info(int slot, char *name) } #endif -#define DAVINCI_PCM_FMTBITS (\ - SNDRV_PCM_FMTBIT_S8 |\ - SNDRV_PCM_FMTBIT_U8 |\ - SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S16_BE |\ - SNDRV_PCM_FMTBIT_U16_LE |\ - SNDRV_PCM_FMTBIT_U16_BE |\ - SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S24_BE |\ - SNDRV_PCM_FMTBIT_U24_LE |\ - SNDRV_PCM_FMTBIT_U24_BE |\ - SNDRV_PCM_FMTBIT_S32_LE |\ - SNDRV_PCM_FMTBIT_S32_BE |\ - SNDRV_PCM_FMTBIT_U32_LE |\ - SNDRV_PCM_FMTBIT_U32_BE) - static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| SNDRV_PCM_INFO_BATCH), - .formats = DAVINCI_PCM_FMTBITS, - .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 384, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, @@ -86,12 +64,6 @@ static struct snd_pcm_hardware pcm_hardware_capture = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH), - .formats = DAVINCI_PCM_FMTBITS, - .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 384, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, -- cgit v0.10.2 From 3cec159cfb3fc69190f3ccdc2d1329c66775529f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:19 +0100 Subject: ASoC: s6000: Don't set unused struct snd_pcm_hardware fields MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Glöckner Signed-off-by: Mark Brown diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index d219880..fb8461e 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -33,13 +33,6 @@ static struct snd_pcm_hardware s6000_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), - .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), - .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000), - .rate_min = 0, - .rate_max = 1562500, - .channels_min = 2, - .channels_max = 8, .buffer_bytes_max = 0x7ffffff0, .period_bytes_min = 16, .period_bytes_max = 0xfffff0, -- cgit v0.10.2 From 60e21d2873440fc005c88970960bed678138217e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:17 +0100 Subject: ASoC: omap: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b8fa986..07b8b7b 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -45,8 +45,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, -- cgit v0.10.2 From 8cb7a36eb3a80cd58353e351b7b4370d9a9f0c14 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 30 Dec 2013 10:37:48 +0530 Subject: ASoC: mcbsp: Trivial cleanup in asoc-ti-mcbsp.h Commit 2203747c9771 ("ARM: omap: move platform_data definitions") moved the file to the current location but forgot to remove the pointer to its previous location. Clean it up. While at it also change the header file protection macros appropriately. Signed-off-by: Sachin Kamat Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/include/linux/platform_data/asoc-ti-mcbsp.h b/include/linux/platform_data/asoc-ti-mcbsp.h index c78d90b..3c73c04 100644 --- a/include/linux/platform_data/asoc-ti-mcbsp.h +++ b/include/linux/platform_data/asoc-ti-mcbsp.h @@ -1,6 +1,4 @@ /* - * arch/arm/plat-omap/include/mach/mcbsp.h - * * Defines for Multi-Channel Buffered Serial Port * * Copyright (C) 2002 RidgeRun, Inc. @@ -21,8 +19,8 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#ifndef __ASM_ARCH_OMAP_MCBSP_H -#define __ASM_ARCH_OMAP_MCBSP_H +#ifndef __ASOC_TI_MCBSP_H +#define __ASOC_TI_MCBSP_H #include #include -- cgit v0.10.2 From e6dc12d7198eddba2e3e7a13feab5c7edde7ba1d Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 25 Dec 2013 11:20:14 +0800 Subject: ASoC: fsl_sai: Move the global registers setting to _dai_probe() Because we cannot make sure which one of _dai_fmt() and _dai_sysclk() will be firstly called. So move the RCSR/TCSR and TCR1/RCR1's initialization to _dai_probe(), and this can make sure that before any of {T,R}CR{1~5} register to be set the RCSR/TCSR's RE/TE bit has been cleared for the hareware limitation. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b72132f..596aabb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -100,11 +100,6 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (ret) return ret; - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { @@ -351,6 +346,18 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); + int ret; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); + sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); + sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); + sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); + + clk_disable_unprepare(sai->clk); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); -- cgit v0.10.2 From 61c66c60c75bd7f9650e27fb0a017bc39f2629a2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 26 Dec 2013 11:41:29 +0530 Subject: ASoC: samsung: Trivial cleanups in header files commit 5d229ce569be ("ASoC: samsung: move plat/ headers to local directory") moved the header files but forgot to clean the pointers to their old locaton. Remove them now. Signed-off-by: Sachin Kamat Reviewed-by: Jingoo Han Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/regs-ac97.h b/sound/soc/samsung/regs-ac97.h index c3878f7..a71be45 100644 --- a/sound/soc/samsung/regs-ac97.h +++ b/sound/soc/samsung/regs-ac97.h @@ -1,5 +1,4 @@ -/* arch/arm/mach-s3c2410/include/mach/regs-ac97.h - * +/* * Copyright (c) 2006 Simtec Electronics * http://www.simtec.co.uk/products/SWLINUX/ * @@ -10,8 +9,8 @@ * S3C2440 AC97 Controller */ -#ifndef __ASM_ARCH_REGS_AC97_H -#define __ASM_ARCH_REGS_AC97_H __FILE__ +#ifndef __SAMSUNG_REGS_AC97_H__ +#define __SAMSUNG_REGS_AC97_H__ #define S3C_AC97_GLBCTRL (0x00) @@ -64,4 +63,4 @@ #define S3C_AC97_PCM_DATA (0x18) #define S3C_AC97_MIC_DATA (0x1C) -#endif /* __ASM_ARCH_REGS_AC97_H */ +#endif /* __SAMSUNG_REGS_AC97_H__ */ diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h index a18d35e..dc6cbbe 100644 --- a/sound/soc/samsung/regs-iis.h +++ b/sound/soc/samsung/regs-iis.h @@ -1,5 +1,4 @@ -/* arch/arm/plat-samsung/include/plat/regs-iis.h - * +/* * Copyright (c) 2003 Simtec Electronics * http://www.simtec.co.uk/products/SWLINUX/ * @@ -10,8 +9,8 @@ * S3C2410 IIS register definition */ -#ifndef __ASM_ARCH_REGS_IIS_H -#define __ASM_ARCH_REGS_IIS_H +#ifndef __SAMSUNG_REGS_IIS_H__ +#define __SAMSUNG_REGS_IIS_H__ #define S3C2410_IISCON (0x00) @@ -67,4 +66,4 @@ #define S3C2410_IISFIFO (0x10) -#endif /* __ASM_ARCH_REGS_IIS_H */ +#endif /* __SAMSUNG_REGS_IIS_H__ */ -- cgit v0.10.2 From e5d0fa9c3ec59a40e0285d96b65b7f62875acd42 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 25 Dec 2013 12:40:04 +0800 Subject: ASoC: fsl_sai: Add disable operation for the corresponding data channel. Enables/Disables the corresponding data channel for tx/rx operation. A channel must be enabled before its FIFO is accessed, and then disable it when tx/rx is stopped or idle. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 596aabb..af80246 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -124,20 +124,17 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; + u32 val_cr2, val_cr4, reg_cr2, reg_cr4; if (fsl_dir == FSL_FMT_TRANSMITTER) { reg_cr2 = FSL_SAI_TCR2; - reg_cr3 = FSL_SAI_TCR3; reg_cr4 = FSL_SAI_TCR4; } else { reg_cr2 = FSL_SAI_RCR2; - reg_cr3 = FSL_SAI_RCR3; reg_cr4 = FSL_SAI_RCR4; } val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr3 = sai_readl(sai, sai->base + reg_cr3); val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) @@ -188,13 +185,10 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - val_cr3 |= FSL_SAI_CR3_TRCE; - if (fsl_dir == FSL_FMT_RECEIVER) val_cr2 |= FSL_SAI_CR2_SYNC; sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr3, sai->base + reg_cr3); sai_writel(sai, val_cr4, sai->base + reg_cr4); return 0; @@ -278,7 +272,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int tcsr, rcsr; + u32 tcsr, rcsr, val_cr3, reg_cr3; tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); @@ -286,17 +280,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; rcsr &= ~FSL_SAI_CSR_FRDE; + reg_cr3 = FSL_SAI_TCR3; } else { rcsr |= FSL_SAI_CSR_FRDE; tcsr &= ~FSL_SAI_CSR_FRDE; + reg_cr3 = FSL_SAI_RCR3; } + val_cr3 = sai_readl(sai, sai->base + reg_cr3); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; + val_cr3 |= FSL_SAI_CR3_TRCE; + + sai_writel(sai, val_cr3, sai->base + reg_cr3); sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); break; @@ -308,8 +309,12 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr &= ~FSL_SAI_CSR_TERE; rcsr &= ~FSL_SAI_CSR_TERE; } + + val_cr3 &= ~FSL_SAI_CR3_TRCE; + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + sai_writel(sai, val_cr3, sai->base + reg_cr3); break; default: return -EINVAL; -- cgit v0.10.2 From 8c5178fca4ce5a57711ea14b807648e19b105d0e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Dec 2013 12:24:28 +0000 Subject: ALSA: Add params_width() helpers Add helpers for obtaining the width of a format directly from params since this is expected to become a common operation in ASoC. Signed-off-by: Mark Brown Reviewed-by: Takashi Iwai diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 37ae12e..6b1c78f 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -354,4 +354,16 @@ params_period_bytes(const struct snd_pcm_hw_params *p) params_channels(p)) / 8; } +static inline int +params_width(const struct snd_pcm_hw_params *p) +{ + return snd_pcm_format_width(params_format(p)); +} + +static inline int +params_physical_width(const struct snd_pcm_hw_params *p) +{ + return snd_pcm_format_physical_width(params_format(p)); +} + #endif /* __SOUND_PCM_PARAMS_H */ -- cgit v0.10.2 From 328089a47112a4fc06071e2003ecd75cc6d31029 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:20 +0100 Subject: ASoC: samsung: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index ee23194..dc09b71 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -35,12 +35,6 @@ static const struct snd_pcm_hardware dma_hardware = { SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = 128*1024, .period_bytes_min = PAGE_SIZE, .period_bytes_max = PAGE_SIZE*2, diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index e4f318f..3d5cf15 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -35,14 +35,6 @@ static const struct snd_pcm_hardware idma_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_U24_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = MAX_IDMA_BUFFER, .period_bytes_min = 128, .period_bytes_max = MAX_IDMA_PERIOD, -- cgit v0.10.2 From 38136bde7691bdafa91c2320e014913aec6dbe6b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:23 +0100 Subject: ASoC: txx9: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index fbd077f..f0829de 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -40,11 +40,6 @@ static const struct snd_pcm_hardware txx9aclc_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_PAUSE, -#ifdef __BIG_ENDIAN - .formats = SNDRV_PCM_FMTBIT_S16_BE, -#else - .formats = SNDRV_PCM_FMTBIT_S16_LE, -#endif .period_bytes_min = 1024, .period_bytes_max = 8 * 1024, .periods_min = 2, -- cgit v0.10.2 From 323702b4e06dfd1a4ee6cee5834a889b9663cccf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:18 +0100 Subject: ASoC: mmp: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 682ee52..5e8d813 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -36,14 +36,9 @@ struct mmp_dma_data { SNDRV_PCM_INFO_PAUSE | \ SNDRV_PCM_INFO_RESUME) -#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - static struct snd_pcm_hardware mmp_pcm_hardware[] = { { .info = MMP_PCM_INFO, - .formats = MMP_PCM_FORMATS, .period_bytes_min = 1024, .period_bytes_max = 2048, .periods_min = 2, @@ -53,7 +48,6 @@ static struct snd_pcm_hardware mmp_pcm_hardware[] = { }, { .info = MMP_PCM_INFO, - .formats = MMP_PCM_FORMATS, .period_bytes_min = 1024, .period_bytes_max = 2048, .periods_min = 2, -- cgit v0.10.2 From 111bd7b18e13c66552e9a672000eeacacd414a65 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:11 +0100 Subject: ASoC: ep93xx: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index ca6698d..5f66447 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -30,15 +30,6 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - - .rates = SNDRV_PCM_RATE_8000_192000, - .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_192000, - - .formats = (SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), - .buffer_bytes_max = 131072, .period_bytes_min = 32, .period_bytes_max = 32768, -- cgit v0.10.2 From 496a39d9ec238569fac6daceac8f5420c5edc2f1 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 31 Dec 2013 15:33:21 +0800 Subject: ASoC: fsl_sai: Fix one bug for hardware limitation. This is maybe one bug or a limitation of the hardware that the {T,R}CR2's Synchronous Mode bits must be set as late as possible, or the SAI device maybe hanged up, and there has not any explaination about this limitation in the SAI Data Sheet. And the {T,R}CR2's Synchronous Mode bits must be set at the same time whether for Tx or Rx stream. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index af80246..2ece147 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -145,7 +145,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: val_cr4 |= FSL_SAI_CR4_FSE; - val_cr4 |= FSL_SAI_CR4_FSP; break; default: return -EINVAL; @@ -185,9 +184,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - if (fsl_dir == FSL_FMT_RECEIVER) - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + reg_cr2); sai_writel(sai, val_cr4, sai->base + reg_cr4); @@ -253,6 +249,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); else @@ -272,7 +269,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr3, reg_cr3; + u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3; + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); + val_cr2 &= ~FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); + val_cr2 |= FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); -- cgit v0.10.2 From 72aa62bed3ea30635156fad95f123a0b665072bf Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 31 Dec 2013 15:33:22 +0800 Subject: ASoC: fsl_sai: fix the endianess for SAI fifo data. Revert the SAI's endianess for fifo data to/from DMA engine. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2ece147..5d38a67 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -138,9 +138,9 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) - val_cr4 |= FSL_SAI_CR4_MF; - else val_cr4 &= ~FSL_SAI_CR4_MF; + else + val_cr4 |= FSL_SAI_CR4_MF; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -251,9 +251,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) - val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); - else val_cr5 |= FSL_SAI_CR5_FBT(0); + else + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); -- cgit v0.10.2 From 192043cf608909eb5728a5fd68f5234b90d9415b Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 27 Nov 2013 18:05:10 +0800 Subject: ASoC: atmel: sam9x5_wm8731: remove platform_set_drvdata When call snd_soc_register_card, it will set driver data to this device through dev_set_drvdata, then in driver, no need to call platform_set_drvdata again, so remove it. Signed-off-by: Bo Shen Signed-off-by: Mark Brown diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38..6f4e812 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -153,8 +153,6 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) of_node_put(codec_np); of_node_put(cpu_np); - platform_set_drvdata(pdev, card); - ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, -- cgit v0.10.2 From d4c22094b256a7327346738721b89a78d4680b08 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 23 Dec 2013 12:57:01 +0800 Subject: ASoC: simple-card: Add DAPM routes parse from device tree Parses a simple DAPM route table from device tree. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 769a346f..2ee80c76 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -9,8 +9,13 @@ Required properties: Optional properties: - simple-audio-card,format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- simple-audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. + Required subnodes: - simple-audio-card,cpu : CPU sub-node @@ -38,6 +43,10 @@ Example: sound { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; + simple-audio-routing = + "MIC_IN", "Mic Jack", + "Headphone Jack", "HP_OUT", + "Ext Spk", "LINE_OUT"; simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3d190d0..6230efb 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,12 +116,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct device_node *np; char *name; - int ret = 0; + int ret; /* get CPU/CODEC common format via simple-audio-card,format */ info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + /* DAPM routes */ + ret = snd_soc_of_parse_audio_routing(&info->snd_card, + "simple-audio-routing"); + if (ret) + return ret; + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -182,6 +188,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; + cinfo->snd_card.dev = &pdev->dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -193,6 +200,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { + cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; } @@ -232,7 +240,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; - cinfo->snd_card.dev = &pdev->dev; return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- cgit v0.10.2 From e874ddead38996ec40c6a6be2347a69fac520126 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 23 Dec 2013 13:24:59 +0800 Subject: ASoC: simple-card: Cleanup __asoc_simple_card_dai_init() ret check The ret parameter is always equal to zero till here. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6230efb..3ba65bb 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,7 +25,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, daifmt |= set->fmt; - if (!ret && daifmt) + if (daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); if (ret == -ENOTSUPP) { -- cgit v0.10.2 From 0f7f3d1f17c2e4d73e449e6acb2007b13813c58e Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 20 Dec 2013 12:40:16 +0200 Subject: ASoC: hdmi-codec: Add devicetree binding with documentation Signed-off-by: Jyri Sarha cc: bcousson@baylibre.com Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/hdmi.txt b/Documentation/devicetree/bindings/sound/hdmi.txt new file mode 100644 index 0000000..31af7bc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/hdmi.txt @@ -0,0 +1,17 @@ +Device-Tree bindings for dummy HDMI codec + +Required properties: + - compatible: should be "linux,hdmi-audio". + +CODEC output pins: + * TX + +CODEC input pins: + * RX + +Example node: + + hdmi_audio: hdmi_audio@0 { + compatible = "linux,hdmi-audio"; + status = "okay"; + }; diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 32797a8..9cb1c7d 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -20,6 +20,7 @@ */ #include #include +#include #define DRV_NAME "hdmi-audio-codec" @@ -60,6 +61,14 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { }; +#ifdef CONFIG_OF +static const struct of_device_id hdmi_audio_codec_ids[] = { + { .compatible = "linux,hdmi-audio", }, + { } +}; +MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids); +#endif + static struct snd_soc_codec_driver hdmi_codec = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), @@ -83,6 +92,7 @@ static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(hdmi_audio_codec_ids), }, .probe = hdmi_codec_probe, -- cgit v0.10.2 From e337853ebb46d012c069ca47ba3ce9f4744305ea Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:26:31 -0800 Subject: ASoC: rsnd: add rsnd_adg_set_ssi_clk() and cleanup adg This patch adds rsnd_adg_set_ssi_clk() to access to AUDIO_CLK_SEL0/1/2, and removes last user of rsnd_write/read/bset which is very low level function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 9430097..55d0394 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -30,41 +30,41 @@ struct rsnd_adg { i++, (pos) = adg->clk[i]) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) -static enum rsnd_reg rsnd_adg_ssi_reg_get(int id) +static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val) { - enum rsnd_reg reg; + int id = rsnd_mod_id(mod); + int shift = (id % 4) * 8; + u32 mask = 0xFF << shift; + + val = val << shift; /* * SSI 8 is not connected to ADG. * it works with SSI 7 */ if (id == 8) - return RSND_REG_MAX; - - if (0 <= id && id <= 3) - reg = RSND_REG_AUDIO_CLK_SEL0; - else if (4 <= id && id <= 7) - reg = RSND_REG_AUDIO_CLK_SEL1; - else - reg = RSND_REG_AUDIO_CLK_SEL2; - - return reg; + return; + + switch (id / 4) { + case 0: + rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val); + break; + case 1: + rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val); + break; + case 2: + rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val); + break; + } } int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - enum rsnd_reg reg; - int id; - /* * "mod" = "ssi" here. * we can get "ssi id" from mod */ - id = rsnd_mod_id(mod); - reg = rsnd_adg_ssi_reg_get(id); - - rsnd_write(priv, mod, reg, 0); + rsnd_adg_set_ssi_clk(mod, 0); return 0; } @@ -75,8 +75,7 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - enum rsnd_reg reg; - int id, shift, i; + int i; u32 data; int sel_table[] = { [CLKA] = 0x1, @@ -125,19 +124,10 @@ found_clock: * This "mod" = "ssi" here. * we can get "ssi id" from mod */ - id = rsnd_mod_id(mod); - reg = rsnd_adg_ssi_reg_get(id); - - dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate); - - /* - * Enable SSIx clock - */ - shift = (id % 4) * 8; + rsnd_adg_set_ssi_clk(mod, data); - rsnd_bset(priv, mod, reg, - 0xFF << shift, - data << shift); + dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", + rsnd_mod_id(mod), i, rate); return 0; } -- cgit v0.10.2 From 729aca51a19f2e2b3404c29b82df61d714150a49 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:26:44 -0800 Subject: ASoC: rsnd: tidyup ssi comment we can check rsnd_ssi_init(), not, rsnd_ssi_start() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 82b04c6..aff5b76 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -209,7 +209,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, /* * this driver is assuming that * system word is 64fs (= 2 x 32bit) - * see rsnd_ssi_start() + * see rsnd_ssi_init() */ main_rate = rate / adg_clk_div_table[i] * 32 * 2 * ssi_clk_mul_table[j]; -- cgit v0.10.2 From 690ef81ebe02a43991b0fcb418d77b8420346cfd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:27:03 -0800 Subject: ASoC: rsnd: tidyup register naming Use correct register name which appears in the datasheet Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index d0ab203..862758d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -314,11 +314,11 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c), RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10), - RSND_GEN1_S_REG(gen, SRU, SRC_CTRL, 0xc0), + RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0), RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4), - RSND_GEN1_M_REG(gen, SRU, BUSIF_ADINR, 0x214, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40), RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index bff7b9e..d5c0182 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -36,12 +36,12 @@ enum rsnd_reg { RSND_REG_SRC_TMG_SEL0, RSND_REG_SRC_TMG_SEL1, RSND_REG_SRC_TMG_SEL2, - RSND_REG_SRC_CTRL, + RSND_REG_SRC_ROUTE_CTRL, RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, - RSND_REG_BUSIF_ADINR, RSND_REG_INT_ENABLE, + RSND_REG_SRC_ADINR, /* ADG */ RSND_REG_BRRA, diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 1ab1bce..187f7dc 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -115,7 +115,7 @@ static int rsnd_scu_set_mode(struct rsnd_priv *priv, if (rsnd_is_gen1(priv)) { val = (1 << id); - rsnd_mod_bset(mod, SRC_CTRL, val, val); + rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); } return 0; @@ -141,7 +141,7 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, } rsnd_mod_write(mod, BUSIF_MODE, 1); - rsnd_mod_write(mod, BUSIF_ADINR, adinr); + rsnd_mod_write(mod, SRC_ADINR, adinr); return 0; } -- cgit v0.10.2 From 7808aa30d6cf366e5f627dcbf7c84f9dc6e602ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:27:19 -0800 Subject: ASoC: rsnd: make sure variable name for 44.1kHz/48kHz This driver is assuming that RBGA is used as source clock of 44.1kHz category, and RBGB is used as source clock of 48kHz category. This patch clarifies the variable name. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 55d0394..2e71a7b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -19,8 +19,8 @@ struct rsnd_adg { struct clk *clk[CLKMAX]; - int rate_of_441khz_div_6; - int rate_of_48khz_div_6; + int rbga_rate_for_441khz_div_6; /* RBGA */ + int rbgb_rate_for_48khz_div_6; /* RBGB */ u32 ckr; }; @@ -101,12 +101,12 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) /* * find 1/6 clock from BRGA/BRGB */ - if (rate == adg->rate_of_441khz_div_6) { + if (rate == adg->rbga_rate_for_441khz_div_6) { data = 0x10; goto found_clock; } - if (rate == adg->rate_of_48khz_div_6) { + if (rate == adg->rbgb_rate_for_48khz_div_6) { data = 0x20; goto found_clock; } @@ -156,8 +156,8 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) * rsnd_adg_ssi_clk_try_start() */ ckr = 0; - adg->rate_of_441khz_div_6 = 0; - adg->rate_of_48khz_div_6 = 0; + adg->rbga_rate_for_441khz_div_6 = 0; + adg->rbgb_rate_for_48khz_div_6 = 0; for_each_rsnd_clk(clk, adg, i) { rate = clk_get_rate(clk); @@ -165,14 +165,14 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) continue; /* RBGA */ - if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) { - adg->rate_of_441khz_div_6 = rate / 6; + if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) { + adg->rbga_rate_for_441khz_div_6 = rate / 6; ckr |= brg_table[i] << 20; } /* RBGB */ - if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) { - adg->rate_of_48khz_div_6 = rate / 6; + if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) { + adg->rbgb_rate_for_48khz_div_6 = rate / 6; ckr |= brg_table[i] << 16; } } -- cgit v0.10.2 From 2582718cb6bd620d37a54db885c75bfe4822db45 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:27:37 -0800 Subject: ASoC: rsnd: route setting is needed only Gen1 Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. rsnd_scu_set_route() is needed only Gen1. This patch renames it to rsnd_scu_set_route_if_gen1() and it adds comment to rsnd_reg member in order to clarify it is used for Gen1. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index d5c0182..a14bc92 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -32,11 +32,11 @@ */ enum rsnd_reg { /* SRU/SCU/SSIU */ - RSND_REG_SRC_ROUTE_SEL, - RSND_REG_SRC_TMG_SEL0, - RSND_REG_SRC_TMG_SEL1, - RSND_REG_SRC_TMG_SEL2, - RSND_REG_SRC_ROUTE_CTRL, + RSND_REG_SRC_ROUTE_SEL, /* for Gen1 */ + RSND_REG_SRC_TMG_SEL0, /* for Gen1 */ + RSND_REG_SRC_TMG_SEL1, /* for Gen1 */ + RSND_REG_SRC_TMG_SEL2, /* for Gen1 */ + RSND_REG_SRC_ROUTE_CTRL, /* for Gen1 */ RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 187f7dc..7642ec5 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -36,7 +36,8 @@ struct rsnd_scu { ((pos) = (struct rsnd_scu *)(priv)->scu + i); \ i++) -static int rsnd_scu_set_route(struct rsnd_priv *priv, +/* Gen1 only */ +static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -174,7 +175,8 @@ static int rsnd_scu_start(struct rsnd_mod *mod, } /* it use DMA transter */ - ret = rsnd_scu_set_route(priv, mod, rdai, io); + + ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io); if (ret < 0) return ret; -- cgit v0.10.2 From af8a478821345fd264fd2294e80f5b0a28a518bc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:04 -0800 Subject: ASoC: rsnd: add rsnd_scu_transfer_start() Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. SRC_CTRL/BUSIF_MODE are used for transfer start. This patch adds rsnd_scu_transfer_start() and merge these Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 7642ec5..3d8b57b1 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -106,22 +106,6 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_set_mode(struct rsnd_priv *priv, - struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - int id = rsnd_mod_id(mod); - u32 val; - - if (rsnd_is_gen1(priv)) { - val = (1 << id); - rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); - } - - return 0; -} - static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, @@ -141,12 +125,29 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, return -EIO; } - rsnd_mod_write(mod, BUSIF_MODE, 1); rsnd_mod_write(mod, SRC_ADINR, adinr); return 0; } +static int rsnd_scu_transfer_start(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + int id = rsnd_mod_id(mod); + u32 val; + + if (rsnd_is_gen1(priv)) { + val = (1 << id); + rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); + } + + rsnd_mod_write(mod, BUSIF_MODE, 1); + + return 0; +} + bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod) { struct rsnd_scu *scu = rsnd_mod_to_scu(mod); @@ -180,11 +181,11 @@ static int rsnd_scu_start(struct rsnd_mod *mod, if (ret < 0) return ret; - ret = rsnd_scu_set_mode(priv, mod, rdai, io); + ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); if (ret < 0) return ret; - ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); + ret = rsnd_scu_transfer_start(priv, mod, rdai, io); if (ret < 0) return ret; -- cgit v0.10.2 From 52ea2a79f440740b57925e729a59337414b4c300 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:19 -0800 Subject: ASoC: rsnd: INT_ENABLE is needed only Gen2 INT_ENABLE is needed only Gen2. rsnd_mod_write() do nothing on Gen1, but it is confusable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a14bc92..3774dfc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -40,7 +40,7 @@ enum rsnd_reg { RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, - RSND_REG_INT_ENABLE, + RSND_REG_INT_ENABLE, /* for Gen2 */ RSND_REG_SRC_ADINR, /* ADG */ diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index aff5b76..01b5cf9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -457,8 +457,9 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; - /* enable PIO interrupt */ - rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000); + /* enable PIO interrupt if gen2 */ + if (rsnd_is_gen2(priv)) + rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000); rsnd_ssi_hw_start(ssi, rdai, io); -- cgit v0.10.2 From 99feec32f26a3c267f89ce48db4bd36650a95f7f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:31 -0800 Subject: ASoC: rsnd: scu cleanup: add rsnd_scu_rate_ctrl() Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. rsnd_scu_set_hpbif() is renamed to rsnd_scu_rate_ctrl(), since its naming doesn't indicate the function meaning. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 3d8b57b1..5f4f572 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -106,7 +106,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, +static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -181,7 +181,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod, if (ret < 0) return ret; - ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); + ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io); if (ret < 0) return ret; -- cgit v0.10.2 From adcf7d5e7605e8134a99d415b7afd13f03c4bf23 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:39 -0800 Subject: ASoC: rsnd: tidyup rsnd_ssi_master_clk_start() parameter Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. ssi clock which is calculated from rsnd_ssi_master_clk_start() should have flexibility since Renesas sound has SRC (= Sampling Rate Converter). But current implementation is using runtime->rate directly. This patch tidyup rsnd_ssi_master_clk_start() parameter as preparation of future SRC support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 01b5cf9..2db9711 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -187,9 +187,10 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, } static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, - unsigned int rate) + struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct device *dev = rsnd_priv_to_dev(priv); int i, j, ret; int adg_clk_div_table[] = { @@ -199,6 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, 1, 2, 4, 8, 16, 6, 12, }; unsigned int main_rate; + unsigned int rate = runtime->rate; /* * Find best clock, and try to start ADG @@ -251,14 +253,10 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, clk_enable(ssi->clk); if (rsnd_rdai_is_clk_master(rdai)) { - struct snd_pcm_runtime *runtime; - - runtime = rsnd_io_to_runtime(io); - if (rsnd_ssi_clk_from_parent(ssi)) rsnd_ssi_hw_start(ssi->parent, rdai, io); else - rsnd_ssi_master_clk_start(ssi, runtime->rate); + rsnd_ssi_master_clk_start(ssi, io); } } -- cgit v0.10.2 From ef749400434cefd14fe02fe3de9e9f0125b2256d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:51 -0800 Subject: ASoC: rsnd: add SRC (Sampling Rate Converter) support This patch adds SRC support to Renesas sound driver. SRC converts sampling rate between codec <-> cpu. It needs special codec chip, or very simple DA/AD converter to use it. This patch was tested via ak4554 codec, and supports Gen1 only at this point. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index a818ff7..e147498 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -58,6 +58,7 @@ struct rsnd_ssi_platform_info { struct rsnd_scu_platform_info { u32 flags; + u32 convert_rate; /* sampling rate convert */ }; /* diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 2e71a7b..a53235c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -30,6 +30,79 @@ struct rsnd_adg { i++, (pos) = adg->clk[i]) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) +static int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv, + struct rsnd_mod *mod, + unsigned int src_rate, + unsigned int dst_rate) +{ + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int idx, sel, div, shift; + u32 mask, val; + int id = rsnd_mod_id(mod); + unsigned int sel_rate [] = { + clk_get_rate(adg->clk[CLKA]), /* 000: CLKA */ + clk_get_rate(adg->clk[CLKB]), /* 001: CLKB */ + clk_get_rate(adg->clk[CLKC]), /* 010: CLKC */ + 0, /* 011: MLBCLK (not used) */ + adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */ + adg->rbgb_rate_for_48khz_div_6, /* 101: RBGB */ + }; + + /* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */ + for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) { + for (div = 128, idx = 0; + div <= 2048; + div *= 2, idx++) { + if (src_rate == sel_rate[sel] / div) { + val = (idx << 4) | sel; + goto find_rate; + } + } + } + dev_err(dev, "can't find convert src clk\n"); + return -EINVAL; + +find_rate: + shift = (id % 4) * 8; + mask = 0xFF << shift; + val = val << shift; + + dev_dbg(dev, "adg convert src clk = %02x\n", val); + + switch (id / 4) { + case 0: + rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val); + break; + case 1: + rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val); + break; + case 2: + rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val); + break; + } + + /* + * Gen1 doesn't need dst_rate settings, + * since it uses SSI WS pin. + * see also rsnd_src_set_route_if_gen1() + */ + + return 0; +} + +int rsnd_adg_set_convert_clk(struct rsnd_priv *priv, + struct rsnd_mod *mod, + unsigned int src_rate, + unsigned int dst_rate) +{ + if (rsnd_is_gen1(priv)) + return rsnd_adg_set_convert_clk_gen1(priv, mod, + src_rate, dst_rate); + + return -EINVAL; +} + static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val) { int id = rsnd_mod_id(mod); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 862758d..add088b 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -318,13 +318,23 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4), + RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8), + RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40), RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_IFSCR, 0x21c, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_IFSVR, 0x220, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_SRCCR, 0x224, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_MNFSR, 0x228, 0x40), RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3774dfc..4ca66cd 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -41,7 +41,14 @@ enum rsnd_reg { RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, RSND_REG_INT_ENABLE, /* for Gen2 */ + RSND_REG_SRC_ROUTE_MODE0, + RSND_REG_SRC_SWRSR, + RSND_REG_SRC_SRCIR, RSND_REG_SRC_ADINR, + RSND_REG_SRC_IFSCR, + RSND_REG_SRC_IFSVR, + RSND_REG_SRC_SRCCR, + RSND_REG_SRC_MNFSR, /* ADG */ RSND_REG_BRRA, @@ -50,6 +57,9 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AUDIO_CLK_SEL3, /* for Gen1 */ + RSND_REG_AUDIO_CLK_SEL4, /* for Gen1 */ + RSND_REG_AUDIO_CLK_SEL5, /* for Gen1 */ /* SSI */ RSND_REG_SSICR, @@ -227,6 +237,10 @@ int rsnd_adg_probe(struct platform_device *pdev, struct rsnd_priv *priv); void rsnd_adg_remove(struct platform_device *pdev, struct rsnd_priv *priv); +int rsnd_adg_set_convert_clk(struct rsnd_priv *priv, + struct rsnd_mod *mod, + unsigned int src_rate, + unsigned int dst_rate); /* * R-Car sound priv @@ -280,6 +294,10 @@ void rsnd_scu_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod); +unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv, + struct rsnd_mod *ssi_mod, + struct snd_pcm_runtime *runtime); + #define rsnd_scu_nr(priv) ((priv)->scu_nr) /* diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 5f4f572..1406dd8 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -13,9 +13,13 @@ struct rsnd_scu { struct rsnd_scu_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + struct clk *clk; }; #define rsnd_scu_mode_flags(p) ((p)->info->flags) +#define rsnd_scu_convert_rate(p) ((p)->info->convert_rate) + +#define RSND_SCU_NAME_SIZE 16 /* * ADINR @@ -26,6 +30,15 @@ struct rsnd_scu { #define OTBL_18 (6 << 16) #define OTBL_16 (8 << 16) +/* + * image of SRC (Sampling Rate Converter) + * + * 96kHz <-> +-----+ 48kHz +-----+ 48kHz +-------+ + * 48kHz <-> | SRC | <------> | SSI | <-----> | codec | + * 44.1kHz <-> +-----+ +-----+ +-------+ + * ... + * + */ #define rsnd_mod_to_scu(_mod) \ container_of((_mod), struct rsnd_scu, mod) @@ -56,7 +69,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, { 0x3, 28, }, /* 7 */ { 0x3, 30, }, /* 8 */ }; - + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); u32 mask; u32 val; int shift; @@ -86,9 +99,18 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, */ shift = (id % 4) * 8; mask = 0x1F << shift; - if (8 == id) /* SRU8 is very special */ + + /* + * ADG is used as source clock if SRC was used, + * then, SSI WS is used as destination clock. + * SSI WS is used as source clock if SRC is not used + * (when playback, source/destination become reverse when capture) + */ + if (rsnd_scu_convert_rate(scu)) /* use ADG */ + val = 0; + else if (8 == id) /* use SSI WS, but SRU8 is special */ val = id << shift; - else + else /* use SSI WS */ val = (id + 1) << shift; switch (id / 4) { @@ -106,14 +128,45 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv, +unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv, + struct rsnd_mod *ssi_mod, + struct snd_pcm_runtime *runtime) +{ + struct rsnd_scu *scu; + unsigned int rate; + + /* this function is assuming SSI id = SCU id here */ + scu = rsnd_mod_to_scu(rsnd_scu_mod_get(priv, rsnd_mod_id(ssi_mod))); + + /* + * return convert rate if SRC is used, + * otherwise, return runtime->rate as usual + */ + rate = rsnd_scu_convert_rate(scu); + if (!rate) + rate = runtime->rate; + + return rate; +} + +static int rsnd_scu_convert_rate_ctrl(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + u32 convert_rate = rsnd_scu_convert_rate(scu); u32 adinr = runtime->channels; + /* set/clear soft reset */ + rsnd_mod_write(mod, SRC_SWRSR, 0); + rsnd_mod_write(mod, SRC_SWRSR, 1); + + /* Initialize the operation of the SRC internal circuits */ + rsnd_mod_write(mod, SRC_SRCIR, 1); + + /* Set channel number and output bit length */ switch (runtime->sample_bits) { case 16: adinr |= OTBL_16; @@ -124,9 +177,42 @@ static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv, default: return -EIO; } - rsnd_mod_write(mod, SRC_ADINR, adinr); + if (convert_rate) { + u32 fsrate = 0x0400000 / convert_rate * runtime->rate; + int ret; + + /* Enable the initial value of IFS */ + rsnd_mod_write(mod, SRC_IFSCR, 1); + + /* Set initial value of IFS */ + rsnd_mod_write(mod, SRC_IFSVR, fsrate); + + /* Select SRC mode (fixed value) */ + rsnd_mod_write(mod, SRC_SRCCR, 0x00010110); + + /* Set the restriction value of the FS ratio (98%) */ + rsnd_mod_write(mod, SRC_MNFSR, fsrate / 100 * 98); + + if (rsnd_is_gen1(priv)) { + /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */ + } + + /* set convert clock */ + ret = rsnd_adg_set_convert_clk(priv, mod, + runtime->rate, + convert_rate); + if (ret < 0) + return ret; + } + + /* Cancel the initialization and operate the SRC function */ + rsnd_mod_write(mod, SRC_SRCIR, 0); + + /* use DMA transfer */ + rsnd_mod_write(mod, BUSIF_MODE, 1); + return 0; } @@ -135,6 +221,7 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); int id = rsnd_mod_id(mod); u32 val; @@ -143,7 +230,28 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv, rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); } - rsnd_mod_write(mod, BUSIF_MODE, 1); + if (rsnd_scu_convert_rate(scu)) + rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + + return 0; +} + +static int rsnd_scu_transfer_stop(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + int id = rsnd_mod_id(mod); + u32 mask; + + if (rsnd_is_gen1(priv)) { + mask = (1 << id); + rsnd_mod_bset(mod, SRC_ROUTE_CTRL, mask, 0); + } + + if (rsnd_scu_convert_rate(scu)) + rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0); return 0; } @@ -161,6 +269,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); struct device *dev = rsnd_priv_to_dev(priv); int ret; @@ -175,13 +284,15 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } + clk_enable(scu->clk); + /* it use DMA transter */ ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io); if (ret < 0) return ret; - ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io); + ret = rsnd_scu_convert_rate_ctrl(priv, mod, rdai, io); if (ret < 0) return ret; @@ -194,9 +305,27 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } +static int rsnd_scu_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + + if (!rsnd_scu_hpbif_is_enable(mod)) + return 0; + + rsnd_scu_transfer_stop(priv, mod, rdai, io); + + clk_disable(scu->clk); + + return 0; +} + static struct rsnd_mod_ops rsnd_scu_ops = { .name = "scu", .start = rsnd_scu_start, + .stop = rsnd_scu_stop, }; struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id) @@ -212,6 +341,8 @@ int rsnd_scu_probe(struct platform_device *pdev, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_scu *scu; + struct clk *clk; + char name[RSND_SCU_NAME_SIZE]; int i, nr; /* @@ -228,9 +359,16 @@ int rsnd_scu_probe(struct platform_device *pdev, priv->scu = scu; for_each_rsnd_scu(scu, priv, i) { + snprintf(name, RSND_SCU_NAME_SIZE, "scu.%d", i); + + clk = devm_clk_get(dev, name); + if (IS_ERR(clk)) + return PTR_ERR(clk); + rsnd_mod_init(priv, &scu->mod, &rsnd_scu_ops, i); scu->info = &info->scu_info[i]; + scu->clk = clk; dev_dbg(dev, "SCU%d probed\n", i); } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2db9711..b7cd06b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -200,7 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, 1, 2, 4, 8, 16, 6, 12, }; unsigned int main_rate; - unsigned int rate = runtime->rate; + unsigned int rate = rsnd_scu_get_ssi_rate(priv, &ssi->mod, runtime); /* * Find best clock, and try to start ADG -- cgit v0.10.2 From 71467e46414d3bab220de77d3d085be0c0aa03e1 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 23 Dec 2013 15:25:38 +0800 Subject: ASoC: simple-card: Add device's module clock selection. Try to get the device's module clock if the dt has no clocks and system-clock-frequency properties. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ba65bb..58c217e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -90,14 +90,29 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * dai->sysclk come from * "clocks = <&xxx>" (if system has common clock) * or "system-clock-frequency = " + * or device's module clock. */ - clk = of_clk_get(np, 0); - if (IS_ERR(clk)) + if (of_property_read_bool(np, "clocks")) { + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + + dai->sysclk = clk_get_rate(clk); + } else if (of_property_read_bool(np, "system-clock-frequency")) { of_property_read_u32(np, "system-clock-frequency", &dai->sysclk); - else + } else { + clk = of_clk_get(*node, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + dai->sysclk = clk_get_rate(clk); + } ret = 0; -- cgit v0.10.2 From f60e5473e6788f93849a61198bec4e02fea31e51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Dec 2013 11:42:55 +0100 Subject: ASoC: ssm2518: Fix off-by-one error by ffs() ffs() returns the bit position from 1, while the ssm2158 driver code assumes it being 0-based. Also, the bit mask computation of the two channel slots are incorrect; it must have worked just casually. Signed-off-by: Takashi Iwai Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 95aed55..cc8debc 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -549,13 +549,13 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, right_slot = 0; } else { /* We assume the left channel < right channel */ - left_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); if (tx_mask == 0) { right_slot = left_slot; } else { - right_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } } -- cgit v0.10.2 From c097d5fdf3b51cdb2521c3cffab0a8cf03c68cc6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Dec 2013 12:41:39 +0000 Subject: ASoC: ad1836: Reject unsupported bit sizes Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 9a92b79..d7c9838 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -179,6 +179,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S32_LE: word_len = AD1836_WORD_LEN_24; break; + default: + return -EINVAL; } regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, -- cgit v0.10.2