From 3228723b0ce0ef6ef6d3f59f282f061430691ab9 Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Mon, 13 Apr 2015 14:20:54 +0800 Subject: ASoC: Intel: Remove invalid kfree of devm allocated data kbuild robot reports following warning: "sound/soc/intel/haswell/sst-haswell-ipc.c:2204:1-6: WARNING: invalid free of devm_ allocated data" As julia explains to me, the memory allocated with devm_kalloc is freed automatically on failure of a probe function. So this kfree should be removed otherwise the double free will be got in error handler path. Signed-off-by: Jin Yao Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 344a1e9..324eceb 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2201,7 +2201,6 @@ dma_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(hsw); return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); -- cgit v0.10.2 From 8c359a9f36796603240863c766a9704e2ad9aa4c Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 17 Apr 2015 22:53:33 +0530 Subject: ASoC: intel - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN we have defined SNDRV_CTL_ELEM_ID_NAME_MAXLEN as size of name array so use this define instead of numeric value Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index daecc58..c55f76a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -695,7 +695,7 @@ struct sst_gain_mixer_control { u16 module_id; u16 pipe_id; u16 task_id; - char pname[44]; + char pname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; struct snd_soc_dapm_widget *w; }; -- cgit v0.10.2 From 044d9601a9dd11ff0e3173ebe34fd30434bd0beb Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Tue, 21 Apr 2015 16:36:00 -0700 Subject: ASoC: Intel: Add support rt5650 in sst driver Added entry in sst driver to support rt5650 codec for intel Braswell platform. Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 05f6930..fc02a48 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -354,6 +354,8 @@ static struct sst_machines sst_acpi_chv[] = { &chv_platform_data }, {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, + {"10EC5650", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", + &chv_platform_data }, {}, }; -- cgit v0.10.2 From cde7fbfc8a2987796fb647e574242fa4bc5430f0 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Fri, 1 May 2015 11:42:02 -0700 Subject: ASoC: Intel: Add support max98090 in sst driver Added entry in sst driver to support max98090 codec for intel Braswell platform. Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index fc02a48..bb19b58 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -356,6 +356,8 @@ static struct sst_machines sst_acpi_chv[] = { &chv_platform_data }, {"10EC5650", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, + {"193C9890", "cht-bsw", "cht-bsw-max98090", NULL, + "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; -- cgit v0.10.2 From 17119a4657066ccefd9a530ab1b07073d97776f8 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Fri, 1 May 2015 11:42:03 -0700 Subject: ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_max98090_ti Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell. This machine driver will support max98090 codec as primary codec. it can also support TI jack detect chip as aux device if platform supports it. Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ee03dbd..01b2b53 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -121,3 +121,15 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with RT5645 audio codec. If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec" + depends on X86_INTEL_LPSS + select SND_SOC_MAX98090 + select SND_SOC_TS3A227E + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with MAX98090 audio codec it also can support TI jack chip as aux device. + If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index f8237f0..cb94895 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -5,6 +5,7 @@ snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o +snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -13,3 +14,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c new file mode 100644 index 0000000..3c518b1 --- /dev/null +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -0,0 +1,320 @@ +/* + * cht-bsw-max98090.c - ASoc Machine driver for Intel Cherryview-based + * platforms Cherrytrail and Braswell, with max98090 & TI codec. + * + * Copyright (C) 2015 Intel Corp + * Author: Fang, Yang A + * This file is modified from cht_bsw_rt5645.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/max98090.h" +#include "../atom/sst-atom-controls.h" +#include "../../codecs/ts3a227e.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "HiFi" + +struct cht_mc_private { + struct snd_soc_jack jack; + bool ts3a227e_present; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_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; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK, + CHT_PLAT_CLK_3_HZ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + int jack_type; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct snd_soc_jack *jack = &ctx->jack; + + /** + * TI supports 4 butons headset detection + * KEY_MEDIA + * KEY_VOICECOMMAND + * KEY_VOLUMEUP + * KEY_VOLUMEDOWN + */ + if (ctx->ts3a227e_present) + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + else + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; + + ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + jack_type, jack, NULL, 0); + + if (ret) { + dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret); + return ret; + } + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + int ret = 0; + unsigned int fmt = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); + if (ret < 0) { + dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret); + return ret; + } + + fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret < 0) { + dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret); + return ret; + } + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static int cht_max98090_headset_init(struct snd_soc_component *component) +{ + struct snd_soc_card *card = component->card; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + + return ts3a227e_enable_jack_detect(component, &ctx->jack); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_aux_dev cht_max98090_headset_dev = { + .name = "Headset Chip", + .init = cht_max98090_headset_init, + .codec_name = "i2c-104C227E:00", +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtmax98090", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .aux_dev = &cht_max98090_headset_dev, + .num_aux_devs = 1, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + bool found = false; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + if (ACPI_SUCCESS(acpi_get_devices( + "104C227E", + snd_acpi_codec_match, + &found, NULL)) && found) { + drv->ts3a227e_present = true; + } else { + /* no need probe TI jack detection chip */ + snd_soc_card_cht.aux_dev = NULL; + snd_soc_card_cht.num_aux_devs = 0; + drv->ts3a227e_present = false; + } + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-max98090", + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-max98090"); -- cgit v0.10.2 From c4ba51ba1c8f8e9dd51f63069eec88580f0e1d01 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Thu, 23 Apr 2015 10:23:02 -0700 Subject: ASoC: Intel: Support rt5650 codec for Cherrytrail & Braswell rt5650 and rt5645 are similar codec so reuse the cht_bsw_rt5645 driver Signed-off-by: Fang, Yang A Acked-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01b2b53..4419d76 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -112,14 +112,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH - tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" depends on X86_INTEL_LPSS select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5645 audio codec. + platforms with RT5645/5650 audio codec. If unsure select "N". config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 20a28b2..7d23ead 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -21,6 +21,7 @@ */ #include +#include #include #include #include @@ -33,9 +34,16 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5645-aif1" +struct cht_acpi_card { + char *codec_id; + int codec_type; + struct snd_soc_card *soc_card; +}; + struct cht_mc_private { struct snd_soc_jack hp_jack; struct snd_soc_jack mic_jack; + struct cht_acpi_card *acpi_card; }; static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) @@ -94,7 +102,7 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD), }; -static const struct snd_soc_dapm_route cht_audio_map[] = { +static const struct snd_soc_dapm_route cht_rt5645_audio_map[] = { {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Int Mic"}, @@ -115,6 +123,27 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"Ext Spk", NULL, "Platform Clock"}, }; +static const struct snd_soc_dapm_route cht_rt5650_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L2", NULL, "Int Mic"}, + {"DMIC R2", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -239,7 +268,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -267,7 +296,7 @@ static struct snd_soc_dai_link cht_dailink[] = { | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, + .nonatomic = true, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, @@ -275,43 +304,85 @@ static struct snd_soc_dai_link cht_dailink[] = { }; /* SoC card */ -static struct snd_soc_card snd_soc_card_cht = { +static struct snd_soc_card snd_soc_card_chtrt5645 = { .name = "chtrt5645", .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), - .dapm_routes = cht_audio_map, - .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .dapm_routes = cht_rt5645_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_rt5645_audio_map), .controls = cht_mc_controls, .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static struct snd_soc_card snd_soc_card_chtrt5650 = { + .name = "chtrt5650", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_rt5650_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_rt5650_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static struct cht_acpi_card snd_soc_cards[] = { + {"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645}, + {"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650}, +}; + +static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + static int snd_cht_mc_probe(struct platform_device *pdev) { int ret_val = 0; + int i; struct cht_mc_private *drv; + struct snd_soc_card *card = snd_soc_cards[0].soc_card; + bool found = false; + char codec_name[16]; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); if (!drv) return -ENOMEM; - snd_soc_card_cht.dev = &pdev->dev; - snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { + if (ACPI_SUCCESS(acpi_get_devices( + snd_soc_cards[i].codec_id, + snd_acpi_codec_match, + &found, NULL)) && found) { + dev_dbg(&pdev->dev, + "found codec %s\n", snd_soc_cards[i].codec_id); + card = snd_soc_cards[i].soc_card; + drv->acpi_card = &snd_soc_cards[i]; + break; + } + } + card->dev = &pdev->dev; + sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ + strcpy((char *)card->dai_link[2].codec_name, codec_name); + snd_soc_card_set_drvdata(card, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, &snd_soc_card_cht); + platform_set_drvdata(pdev, card); return ret_val; } static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5645", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; -- cgit v0.10.2 From 26f63c692f012ff665a8fd085a36549fe734f59f Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 4 May 2015 13:43:47 -0700 Subject: ASoC: Intel: Fixed kbuild warnings fix following sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: sparse: >> incorrect type in argument 2 (different base types) sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: expected unsigned int [unsigned] val sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: got restricted snd_pcm_format_t [usertype] Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 3c518b1..1be0794 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -163,9 +163,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } -- cgit v0.10.2 From 673c4f896a10a8df7d09525fe41f5663e0ca1bd4 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Tue, 5 May 2015 16:55:34 -0700 Subject: ASoC: Intel: Enabled button jack for BSW platform with rt5650 codec rt5650 codec supports 4 buttons detections so enabled it Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8f96c21..bdcaf46 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -41,8 +41,7 @@ struct cht_acpi_card { }; struct cht_mc_private { - struct snd_soc_jack hp_jack; - struct snd_soc_jack mic_jack; + struct snd_soc_jack jack; struct cht_acpi_card *acpi_card; }; @@ -179,6 +178,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; + int jack_type; struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dai *codec_dai = runtime->codec_dai; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); @@ -198,23 +198,22 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", - SND_JACK_HEADPHONE, &ctx->hp_jack, - NULL, 0); - if (ret) { - dev_err(runtime->dev, "HP jack creation failed %d\n", ret); - return ret; - } + if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650) + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + else + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", - SND_JACK_MICROPHONE, &ctx->mic_jack, + ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + jack_type, &ctx->jack, NULL, 0); if (ret) { - dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); + dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); return ret; } - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL); + rt5645_set_jack_detect(codec, &ctx->jack, &ctx->jack, &ctx->jack); return ret; } -- cgit v0.10.2 From 0eb93ef04b2641d4140e11d6b1f2f3841edd9a7a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 1 May 2015 18:02:44 +0200 Subject: ASoC: lm4857: Use DAPM demux Use a DAPM auto-disable demux to model the Mode control which affects the routing of the input pin to the output pins. This allows us to remove the custom code for handling the Mode control. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 79ad4cb..dac9165 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -23,11 +23,6 @@ #include #include -struct lm4857 { - struct regmap *regmap; - uint8_t mode; -}; - static const struct reg_default lm4857_default_regs[] = { { 0x0, 0x00 }, { 0x1, 0x00 }, @@ -46,64 +41,33 @@ static const struct reg_default lm4857_default_regs[] = { #define LM4857_WAKEUP 5 #define LM4857_EPGAIN 4 -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - - ucontrol->value.integer.value[0] = lm4857->mode; - - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - uint8_t value = ucontrol->value.integer.value[0]; - - lm4857->mode = value; - - if (codec->dapm.bias_level == SND_SOC_BIAS_ON) - regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6); - - return 1; -} - -static int lm4857_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - - switch (level) { - case SND_SOC_BIAS_ON: - regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, - lm4857->mode + 6); - break; - case SND_SOC_BIAS_STANDBY: - regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0); - break; - default: - break; - } - - return 0; -} +static const unsigned int lm4857_mode_values[] = { + 0, + 6, + 7, + 8, + 9, +}; -static const char *lm4857_mode[] = { +static const char * const lm4857_mode_texts[] = { + "Off", "Earpiece", "Loudspeaker", "Loudspeaker + Headphone", "Headphone", }; -static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode); +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(lm4857_mode_enum, + LM4857_CTRL, 0, 0xf, lm4857_mode_texts, lm4857_mode_values); + +static const struct snd_kcontrol_new lm4857_mode_ctrl = + SOC_DAPM_ENUM("Mode", lm4857_mode_enum); static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN"), + SND_SOC_DAPM_DEMUX("Mode", SND_SOC_NOPM, 0, 0, &lm4857_mode_ctrl), + SND_SOC_DAPM_OUTPUT("LS"), SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_OUTPUT("EP"), @@ -125,24 +89,18 @@ static const struct snd_kcontrol_new lm4857_controls[] = { LM4857_WAKEUP, 1, 0), SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL, LM4857_EPGAIN, 1, 0), - - SOC_ENUM_EXT("Mode", lm4857_mode_enum, - lm4857_get_mode, lm4857_set_mode), }; -/* There is a demux between the input signal and the output signals. - * Currently there is no easy way to model it in ASoC and since it does not make - * much of a difference in practice simply connect the input direclty to the - * outputs. */ static const struct snd_soc_dapm_route lm4857_routes[] = { - {"LS", NULL, "IN"}, - {"HP", NULL, "IN"}, - {"EP", NULL, "IN"}, + { "Mode", NULL, "IN" }, + { "LS", "Loudspeaker", "Mode" }, + { "LS", "Loudspeaker + Headphone", "Mode" }, + { "HP", "Headphone", "Mode" }, + { "HP", "Loudspeaker + Headphone", "Mode" }, + { "EP", "Earpiece", "Mode" }, }; static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { - .set_bias_level = lm4857_set_bias_level, - .controls = lm4857_controls, .num_controls = ARRAY_SIZE(lm4857_controls), .dapm_widgets = lm4857_dapm_widgets, @@ -165,17 +123,11 @@ static const struct regmap_config lm4857_regmap_config = { static int lm4857_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct lm4857 *lm4857; - - lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL); - if (!lm4857) - return -ENOMEM; - - i2c_set_clientdata(i2c, lm4857); + struct regmap *regmap; - lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); - if (IS_ERR(lm4857->regmap)) - return PTR_ERR(lm4857->regmap); + regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); } -- cgit v0.10.2 From 08a1e646bdc1d0e14d2ea19075a916619bafd271 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 1 May 2015 18:02:45 +0200 Subject: ASoC: lm4857: Convert to component The driver does not use any CODEC specific constructs anymore. Convert it to snd_soc_component. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index dac9165..99ffc49 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -100,7 +100,7 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { { "EP", "Earpiece", "Mode" }, }; -static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { +static struct snd_soc_component_driver lm4857_component_driver = { .controls = lm4857_controls, .num_controls = ARRAY_SIZE(lm4857_controls), .dapm_widgets = lm4857_dapm_widgets, @@ -129,13 +129,8 @@ static int lm4857_i2c_probe(struct i2c_client *i2c, if (IS_ERR(regmap)) return PTR_ERR(regmap); - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); -} - -static int lm4857_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; + return devm_snd_soc_register_component(&i2c->dev, + &lm4857_component_driver, NULL, 0); } static const struct i2c_device_id lm4857_i2c_id[] = { @@ -150,7 +145,6 @@ static struct i2c_driver lm4857_i2c_driver = { .owner = THIS_MODULE, }, .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, .id_table = lm4857_i2c_id, }; -- cgit v0.10.2 From 1f2d86f1c0c9283daa8f215cfe465125c81a6fe5 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 6 May 2015 22:06:40 +0530 Subject: ASoC: Intel: add frame and data polarity to ssp config The current ssp configuration was not configuring the frame sync polarity and data polarity. Some codecs do need these different so add them in ssp configuration now Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 90aa5c0..59517b3 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -789,6 +789,8 @@ static const struct sst_ssp_config sst_ssp_configs = { .fs_frequency = SSP_FS_48_KHZ, .active_slot_map = 0xF, .start_delay = 0, + .frame_sync_polarity = SSP_FS_ACTIVE_HIGH, + .data_polarity = 1, }; int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index c55f76a..eea7156 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -562,6 +562,8 @@ struct sst_ssp_config { u8 active_slot_map; u8 start_delay; u16 fs_width; + u8 frame_sync_polarity; + u8 data_polarity; }; struct sst_ssp_cfg { -- cgit v0.10.2 From 5749d70edc2796606dfea3b6b6b5524607634453 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 6 May 2015 22:06:41 +0530 Subject: ASoC: Intel: use local values for ssp configuration So right now SSP configuration is statically coded in the driver. While we would like to keep this configuration intact for the users who are using these defaults, we need to provide a way for users to program it. So create a local value in driver structure which is populate with default value for now Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 59517b3..93c6c8b 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -793,45 +793,52 @@ static const struct sst_ssp_config sst_ssp_configs = { .data_polarity = 1, }; +void sst_fill_ssp_defaults(struct snd_soc_dai *dai) +{ + const struct sst_ssp_config *config; + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); + + config = &sst_ssp_configs; + + ctx->ssp_cmd.selection = config->ssp_id; + ctx->ssp_cmd.nb_bits_per_slots = config->bits_per_slot; + ctx->ssp_cmd.nb_slots = config->slots; + ctx->ssp_cmd.mode = config->ssp_mode | (config->pcm_mode << 1); + ctx->ssp_cmd.duplex = config->duplex; + ctx->ssp_cmd.active_tx_slot_map = config->active_slot_map; + ctx->ssp_cmd.active_rx_slot_map = config->active_slot_map; + ctx->ssp_cmd.frame_sync_frequency = config->fs_frequency; + ctx->ssp_cmd.frame_sync_polarity = config->frame_sync_polarity; + ctx->ssp_cmd.data_polarity = config->data_polarity; + ctx->ssp_cmd.frame_sync_width = config->fs_width; + ctx->ssp_cmd.ssp_protocol = config->ssp_protocol; + ctx->ssp_cmd.start_delay = config->start_delay; + ctx->ssp_cmd.reserved1 = ctx->ssp_cmd.reserved2 = 0xFF; +} + int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) { - struct sst_cmd_sba_hw_set_ssp cmd; struct sst_data *drv = snd_soc_dai_get_drvdata(dai); const struct sst_ssp_config *config; dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id); - SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); - cmd.header.command_id = SBA_HW_SET_SSP; - cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp) + SST_FILL_DEFAULT_DESTINATION(drv->ssp_cmd.header.dst); + drv->ssp_cmd.header.command_id = SBA_HW_SET_SSP; + drv->ssp_cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp) - sizeof(struct sst_dsp_header); config = &sst_ssp_configs; dev_dbg(dai->dev, "ssp_id: %u\n", config->ssp_id); if (enable) - cmd.switch_state = SST_SWITCH_ON; + drv->ssp_cmd.switch_state = SST_SWITCH_ON; else - cmd.switch_state = SST_SWITCH_OFF; - - cmd.selection = config->ssp_id; - cmd.nb_bits_per_slots = config->bits_per_slot; - cmd.nb_slots = config->slots; - cmd.mode = config->ssp_mode | (config->pcm_mode << 1); - cmd.duplex = config->duplex; - cmd.active_tx_slot_map = config->active_slot_map; - cmd.active_rx_slot_map = config->active_slot_map; - cmd.frame_sync_frequency = config->fs_frequency; - cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH; - cmd.data_polarity = 1; - cmd.frame_sync_width = config->fs_width; - cmd.ssp_protocol = config->ssp_protocol; - cmd.start_delay = config->start_delay; - cmd.reserved1 = cmd.reserved2 = 0xFF; + drv->ssp_cmd.switch_state = SST_SWITCH_OFF; return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, - SST_TASK_SBA, 0, &cmd, - sizeof(cmd.header) + cmd.header.length); + SST_TASK_SBA, 0, &drv->ssp_cmd, + sizeof(drv->ssp_cmd.header) + drv->ssp_cmd.header.length); } static int sst_set_be_modules(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/intel/atom/sst-mfld-platform.h b/sound/soc/intel/atom/sst-mfld-platform.h index 9094314..2409b23 100644 --- a/sound/soc/intel/atom/sst-mfld-platform.h +++ b/sound/soc/intel/atom/sst-mfld-platform.h @@ -22,6 +22,7 @@ #define __SST_PLATFORMDRV_H__ #include "sst-mfld-dsp.h" +#include "sst-atom-controls.h" extern struct sst_device *sst; @@ -175,6 +176,7 @@ struct sst_data { struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; struct snd_soc_card *soc_card; + struct sst_cmd_sba_hw_set_ssp ssp_cmd; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); -- cgit v0.10.2 From 711bc9476bfaeba279259978aadcaa826a77e170 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 6 May 2015 22:06:42 +0530 Subject: ASoC: Intel: load hw_defaults in hw_params of ssp be We have the SSP defaults now and we need to load then in hw_params callback of BE SSP DAI ops. Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index eea7156..da13f6f 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -869,4 +869,6 @@ struct sst_enum { SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \ SST_SSP_MUX_ENUM(xreg, xshift, xtexts)) +void sst_fill_ssp_defaults(struct snd_soc_dai *dai); + #endif diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 2fbaf2c..1fb2448 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -434,13 +434,22 @@ static int sst_enable_ssp(struct snd_pcm_substream *substream, if (!dai->active) { ret = sst_handle_vb_timer(dai, true); - if (ret) - return ret; - ret = send_ssp_cmd(dai, dai->name, 1); + sst_fill_ssp_defaults(dai); } return ret; } +static int sst_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + + if (dai->active == 1) + ret = send_ssp_cmd(dai, dai->name, 1); + return ret; +} + static void sst_disable_ssp(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -465,6 +474,7 @@ static struct snd_soc_dai_ops sst_compr_dai_ops = { static struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, + .hw_params = sst_be_hw_params, .shutdown = sst_disable_ssp, }; -- cgit v0.10.2 From 0b44e345495ad97d533461e53a9218de8039d20b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 6 May 2015 22:06:43 +0530 Subject: ASoC: intel: add support for specifying PCM format With this machines can configure the PCM format applied on SSP port using the set_fmt API Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 93c6c8b..e024d98 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -774,8 +774,107 @@ int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable) return ret; } +static int sst_get_frame_sync_polarity(struct snd_soc_dai *dai, + unsigned int fmt) +{ + int format; + + format = fmt & SND_SOC_DAIFMT_INV_MASK; + dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format); + + switch (format) { + case SND_SOC_DAIFMT_NB_NF: + return SSP_FS_ACTIVE_LOW; + case SND_SOC_DAIFMT_NB_IF: + return SSP_FS_ACTIVE_HIGH; + case SND_SOC_DAIFMT_IB_IF: + return SSP_FS_ACTIVE_LOW; + case SND_SOC_DAIFMT_IB_NF: + return SSP_FS_ACTIVE_HIGH; + default: + dev_err(dai->dev, "Invalid frame sync polarity %d\n", format); + } + + return -EINVAL; +} + +static int sst_get_ssp_mode(struct snd_soc_dai *dai, unsigned int fmt) +{ + int format; + + format = (fmt & SND_SOC_DAIFMT_MASTER_MASK); + dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format); + + switch (format) { + case SND_SOC_DAIFMT_CBS_CFS: + return SSP_MODE_MASTER; + case SND_SOC_DAIFMT_CBM_CFM: + return SSP_MODE_SLAVE; + default: + dev_err(dai->dev, "Invalid ssp protocol: %d\n", format); + } + + return -EINVAL; +} + + +int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt) +{ + unsigned int mode; + int fs_polarity; + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); + + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (mode) { + case SND_SOC_DAIFMT_DSP_B: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_PCM; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NETWORK << 1); + ctx->ssp_cmd.start_delay = 0; + ctx->ssp_cmd.data_polarity = 1; + ctx->ssp_cmd.frame_sync_width = 1; + break; + + case SND_SOC_DAIFMT_DSP_A: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_PCM; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NETWORK << 1); + ctx->ssp_cmd.start_delay = 1; + ctx->ssp_cmd.data_polarity = 1; + ctx->ssp_cmd.frame_sync_width = 1; + break; + + case SND_SOC_DAIFMT_I2S: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_I2S; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NORMAL << 1); + ctx->ssp_cmd.start_delay = 1; + ctx->ssp_cmd.data_polarity = 0; + ctx->ssp_cmd.frame_sync_width = ctx->ssp_cmd.nb_bits_per_slots; + break; + + case SND_SOC_DAIFMT_LEFT_J: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_I2S; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NORMAL << 1); + ctx->ssp_cmd.start_delay = 0; + ctx->ssp_cmd.data_polarity = 0; + ctx->ssp_cmd.frame_sync_width = ctx->ssp_cmd.nb_bits_per_slots; + break; + + default: + dev_dbg(dai->dev, "using default ssp configs\n"); + } + + fs_polarity = sst_get_frame_sync_polarity(dai, fmt); + if (fs_polarity < 0) + return fs_polarity; + + ctx->ssp_cmd.frame_sync_polarity = fs_polarity; + + return 0; +} + /** * sst_ssp_config - contains SSP configuration for media UC + * this can be overwritten by set_dai_xxx APIs */ static const struct sst_ssp_config sst_ssp_configs = { .ssp_id = SSP_CODEC, diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index da13f6f..53551a6 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -869,6 +869,7 @@ struct sst_enum { SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \ SST_SSP_MUX_ENUM(xreg, xshift, xtexts)) +int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt); void sst_fill_ssp_defaults(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 1fb2448..580f5e9 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -450,6 +450,20 @@ static int sst_be_hw_params(struct snd_pcm_substream *substream, return ret; } +static int sst_set_format(struct snd_soc_dai *dai, unsigned int fmt) +{ + int ret = 0; + + if (!dai->active) + return 0; + + ret = sst_fill_ssp_config(dai, fmt); + if (ret < 0) + dev_err(dai->dev, "sst_set_format failed..\n"); + + return ret; +} + static void sst_disable_ssp(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -475,6 +489,7 @@ static struct snd_soc_dai_ops sst_compr_dai_ops = { static struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, .hw_params = sst_be_hw_params, + .set_fmt = sst_set_format, .shutdown = sst_disable_ssp, }; -- cgit v0.10.2 From 83f125e2a1a3c7aba9c40016b9d4bec4d43f165d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 6 May 2015 22:06:44 +0530 Subject: ASoC: Intel: add support for configuring TDM slots for SSP With this machines can now configure TDM settings for SSP port using set_tdm_slot API Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index e024d98..61e2409 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -774,6 +774,19 @@ int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable) return ret; } +int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); + + ctx->ssp_cmd.nb_slots = slots; + ctx->ssp_cmd.active_tx_slot_map = tx_mask; + ctx->ssp_cmd.active_rx_slot_map = rx_mask; + ctx->ssp_cmd.nb_bits_per_slots = slot_width; + + return 0; +} + static int sst_get_frame_sync_polarity(struct snd_soc_dai *dai, unsigned int fmt) { diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index 53551a6..93de804 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -869,6 +869,8 @@ struct sst_enum { SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \ SST_SSP_MUX_ENUM(xreg, xshift, xtexts)) +int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width); int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt); void sst_fill_ssp_defaults(struct snd_soc_dai *dai); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 580f5e9..641ebe6 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -464,6 +464,21 @@ static int sst_set_format(struct snd_soc_dai *dai, unsigned int fmt) return ret; } +static int sst_platform_set_ssp_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) { + int ret = 0; + + if (!dai->active) + return ret; + + ret = sst_fill_ssp_slot(dai, tx_mask, rx_mask, slots, slot_width); + if (ret < 0) + dev_err(dai->dev, "sst_fill_ssp_slot failed..%d\n", ret); + + return ret; +} + static void sst_disable_ssp(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -490,6 +505,7 @@ static struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, .hw_params = sst_be_hw_params, .set_fmt = sst_set_format, + .set_tdm_slot = sst_platform_set_ssp_slot, .shutdown = sst_disable_ssp, }; -- cgit v0.10.2 From 40579e0b88580cb8fd53218635ab0afbdb3a4919 Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 7 May 2015 21:29:31 +0200 Subject: ASoC: gtm601: Document GTM601 bindings Add small documentation for GTM601 UMTS modem audio interface. Signed-off-by: Marek Belisko Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt new file mode 100644 index 0000000..5efc8c0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.txt @@ -0,0 +1,13 @@ +GTM601 UMTS modem audio interface CODEC + +This device has no configuration interface. Sample rate is fixed - 8kHz. + +Required properties: + + - compatible : "option,gtm601" + +Example: + +codec: gtm601_codec { + compatible = "option,gtm601"; +}; -- cgit v0.10.2 From 1137e58069ac8ce8df5d691f340b7e184616c84a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 19 May 2015 08:54:27 +0200 Subject: ASoC: sta32x: use devm_gpiod_get_optional for optional reset gpio MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Also there is a variant to find optional gpios that returns NULL if there is no gpio instead of -ENOENT. Make use of both features to simplify the driver. This changes behaviour if gpiod_get returns -ENOSYS which is the case if CONFIG_GPIOLIB is not enabled. This is a good change because without GPIOLIB there is no way to determine if the reset gpio is specified in the device tree. And if it is it must be handled, so erroring out is the right thing to do. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 007a0e3..0111baf 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1096,16 +1096,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- cgit v0.10.2 From 5edf1e06927caba17ffa4489f2d81700cc932969 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 19 May 2015 08:58:09 +0200 Subject: ASoC: max98357a: use flags argument of devm_gpiod_get to set direction MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Use this to simplify the driver. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Signed-off-by: Uwe Kleine-König Acked-by: Kenneth Westfield Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index bf3e933..3a2fda0 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -60,13 +60,12 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) { struct gpio_desc *sdmode; - sdmode = devm_gpiod_get(codec->dev, "sdmode"); + sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); if (IS_ERR(sdmode)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); return PTR_ERR(sdmode); } - gpiod_direction_output(sdmode, 0); snd_soc_codec_set_drvdata(codec, sdmode); return 0; -- cgit v0.10.2 From 0a8ba6eeb6501a77619f49440c85dad14fe9c7a2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 19 May 2015 09:48:08 +0200 Subject: ASoC: rx51: use flags argument of devm_gpiod_get to set direction MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Use this to simplify the driver. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Signed-off-by: Uwe Kleine-König Acked-by: Jarkko Nikula Signed-off-by: Mark Brown diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index c2ddf0f..fded993 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -455,50 +455,36 @@ static int rx51_soc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, pdata); pdata->tvout_selection_gpio = devm_gpiod_get(card->dev, - "tvout-selection"); + "tvout-selection", + GPIOD_OUT_LOW); if (IS_ERR(pdata->tvout_selection_gpio)) { dev_err(card->dev, "could not get tvout selection gpio\n"); return PTR_ERR(pdata->tvout_selection_gpio); } - err = gpiod_direction_output(pdata->tvout_selection_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup tvout selection gpio\n"); - return err; - } - pdata->jack_detection_gpio = devm_gpiod_get(card->dev, - "jack-detection"); + "jack-detection", + GPIOD_ASIS); if (IS_ERR(pdata->jack_detection_gpio)) { dev_err(card->dev, "could not get jack detection gpio\n"); return PTR_ERR(pdata->jack_detection_gpio); } - pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch"); + pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch", + GPIOD_OUT_HIGH); if (IS_ERR(pdata->eci_sw_gpio)) { dev_err(card->dev, "could not get eci switch gpio\n"); return PTR_ERR(pdata->eci_sw_gpio); } - err = gpiod_direction_output(pdata->eci_sw_gpio, 1); - if (err) { - dev_err(card->dev, "could not setup eci switch gpio\n"); - return err; - } - pdata->speaker_amp_gpio = devm_gpiod_get(card->dev, - "speaker-amplifier"); + "speaker-amplifier", + GPIOD_OUT_LOW); if (IS_ERR(pdata->speaker_amp_gpio)) { dev_err(card->dev, "could not get speaker enable gpio\n"); return PTR_ERR(pdata->speaker_amp_gpio); } - err = gpiod_direction_output(pdata->speaker_amp_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup speaker enable gpio\n"); - return err; - } - err = devm_snd_soc_register_card(card->dev, card); if (err) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); -- cgit v0.10.2 From 6022d330a59735adbdcb917d1428a306dbba577b Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:33 +0530 Subject: ASoC: Intel: Create an ops to check for DSP busy Created an ops to check if DSP busy, to avoid using platform specific registers in common IPC. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h index 125ea45..77a3bef 100644 --- a/sound/soc/intel/common/sst-ipc.h +++ b/sound/soc/intel/common/sst-ipc.h @@ -51,6 +51,7 @@ struct sst_plat_ipc_ops { void (*shim_dbg)(struct sst_generic_ipc *, const char *); void (*tx_data_copy)(struct ipc_message *, char *, size_t); u64 (*reply_msg_match)(u64 header, u64 *mask); + bool (*is_dsp_busy)(struct sst_dsp *dsp); }; /* SST generic IPC data */ -- cgit v0.10.2 From 2709bdbc4d7ffae3bcd3e24e214475fcc3d4f77e Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:34 +0530 Subject: ASoC: Intel: Move the busy check to ops for Baytrail Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 1efb33b..799b804 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -679,6 +679,14 @@ static u64 byt_reply_msg_match(u64 header, u64 *mask) return header; } +static bool byt_is_dsp_busy(struct sst_dsp *dsp) +{ + u64 ipcx; + + ipcx = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + return (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)); +} + int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; @@ -699,6 +707,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) ipc->ops.shim_dbg = byt_shim_dbg; ipc->ops.tx_data_copy = byt_tx_data_copy; ipc->ops.reply_msg_match = byt_reply_msg_match; + ipc->ops.is_dsp_busy = byt_is_dsp_busy; err = sst_ipc_init(ipc); if (err != 0) -- cgit v0.10.2 From 40fea92107ce0d7465e52cd7b1a2b7883618ba1b Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:35 +0530 Subject: ASoC: Intel: Move the busy check to ops for HSW Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 324eceb..6304e4b 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2098,6 +2098,14 @@ static u64 hsw_reply_msg_match(u64 header, u64 *mask) return header; } +static bool hsw_is_dsp_busy(struct sst_dsp *dsp) +{ + u64 ipcx; + + ipcx = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + return (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)); +} + int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_hsw_ipc_fw_version version; @@ -2117,6 +2125,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) ipc->ops.shim_dbg = hsw_shim_dbg; ipc->ops.tx_data_copy = hsw_tx_data_copy; ipc->ops.reply_msg_match = hsw_reply_msg_match; + ipc->ops.is_dsp_busy = hsw_is_dsp_busy; ret = sst_ipc_init(ipc); if (ret != 0) -- cgit v0.10.2 From a63faa58bd90477f143f6a9700db91a17593796e Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:36 +0530 Subject: ASoC: Intel: Remove the direct register reference from common ipc Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 4b62a55..a7699f3 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -142,7 +142,6 @@ static void ipc_tx_msgs(struct kthread_work *work) container_of(work, struct sst_generic_ipc, kwork); struct ipc_message *msg; unsigned long flags; - u64 ipcx; spin_lock_irqsave(&ipc->dsp->spinlock, flags); @@ -153,8 +152,8 @@ static void ipc_tx_msgs(struct kthread_work *work) /* if the DSP is busy, we will TX messages after IRQ. * also postpone if we are in the middle of procesing completion irq*/ - ipcx = sst_dsp_shim_read_unlocked(ipc->dsp, SST_IPCX); - if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { + if (ipc->ops.is_dsp_busy && ipc->ops.is_dsp_busy(ipc->dsp)) { + dev_dbg(ipc->dev, "ipc_tx_msgs dsp busy\n"); spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); return; } -- cgit v0.10.2 From 1925e219610d283901b21a4468e86421baa580b8 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:37 +0530 Subject: ASoC: Intel: Allow to configure max size for mailbox data Mailbox size can be different for different platforms. So allow the drivers to configure the size. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h index 77a3bef..7139afd 100644 --- a/sound/soc/intel/common/sst-ipc.h +++ b/sound/soc/intel/common/sst-ipc.h @@ -69,6 +69,8 @@ struct sst_generic_ipc { struct kthread_work kwork; bool pending; struct ipc_message *msg; + int tx_data_max_size; + int rx_data_max_size; struct sst_plat_ipc_ops ops; }; -- cgit v0.10.2 From f99b26f0b4472f4359d123e11530ad43fcd6702d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:38 +0530 Subject: ASoC: Intel: Initialize max mailbox size for baytrail Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 799b804..773a475 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -708,6 +708,8 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) ipc->ops.tx_data_copy = byt_tx_data_copy; ipc->ops.reply_msg_match = byt_reply_msg_match; ipc->ops.is_dsp_busy = byt_is_dsp_busy; + ipc->tx_data_max_size = IPC_MAX_MAILBOX_BYTES; + ipc->rx_data_max_size = IPC_MAX_MAILBOX_BYTES; err = sst_ipc_init(ipc); if (err != 0) -- cgit v0.10.2 From d0e72cc0ac3dcebf0de179ba1dd33a276642c5bb Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:39 +0530 Subject: ASoC: Intel: Initialize max mailbox size for haswell Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 6304e4b..f95f271 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2127,6 +2127,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) ipc->ops.reply_msg_match = hsw_reply_msg_match; ipc->ops.is_dsp_busy = hsw_is_dsp_busy; + ipc->tx_data_max_size = IPC_MAX_MAILBOX_BYTES; + ipc->rx_data_max_size = IPC_MAX_MAILBOX_BYTES; + ret = sst_ipc_init(ipc); if (ret != 0) goto ipc_init_err; -- cgit v0.10.2 From 506c148ee5e1bfb836116353305927ca4c21a23e Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 28 May 2015 22:51:54 +0800 Subject: ASoC: Intel: remove unused function hsw_pcm_free_modules() Remove the unused function hsw_pcm_free_modules() to fix the compling warning: sound/soc/intel/haswell/sst-haswell-pcm.c:923:13: warning: 'sw_pcm_free_modules' defined but not used [-Wunused-function] static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 23ae040..225c04c 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -920,21 +920,6 @@ err: return -ENODEV; } -static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) -{ - struct sst_hsw *hsw = pdata->hsw; - struct hsw_pcm_data *pcm_data; - int i; - - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - sst_hsw_runtime_module_free(pcm_data->runtime); - } - if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { - sst_hsw_runtime_module_free(pdata->runtime_waves); - } -} - static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; -- cgit v0.10.2 From 01f202c7b4b40025f3ea4721c52e7f78545e3b07 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 28 May 2015 14:14:18 +0800 Subject: ASoC: Intel: fix broadwell module removing failed issue In haswell-pcm module unloading, we can't free runtime modules directly, for they may be already freed in runtime suspend. Here add executing suspend call to unload runtime modules, only for status not equal to RPM_SUSPEND, to fix broadwell module removing failed issue. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 225c04c..1557e37 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1103,8 +1103,10 @@ static int hsw_pcm_remove(struct snd_soc_platform *platform) snd_soc_platform_get_drvdata(platform); int i; + /* execute a suspend call to unload all FW resources */ + if (!pm_runtime_status_suspended(platform->dev)) + pm_runtime_put_sync_suspend(platform->dev); pm_runtime_disable(platform->dev); - hsw_pcm_free_modules(priv_data); for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { if (hsw_dais[i].playback.channels_min) -- cgit v0.10.2 From 616268292b274d57aa02d20815f68ad2bd7e1cf7 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Sat, 30 May 2015 15:58:47 +0800 Subject: ASoC: Intel: don't need compress offload for broadwell We don't need compress offload feature for broadwell broadwell machine, here remove the non exist dependency. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4419d76..791953f 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -79,7 +79,6 @@ config SND_SOC_INTEL_BROADWELL_MACH depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL - select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell -- cgit v0.10.2 From a209d322dc803d2bb0c92fe1d0c703ddabae6f28 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Sat, 30 May 2015 22:33:56 +0800 Subject: ASoC: intel: Revert "ASoC: Intel: remove unused function hsw_pcm_free_modules()" This reverts commit 506c148ee5e1bfb836116353305927ca4c21a23e. We still need this hsw_pcm_free_modules(), we plan to remove the runtime modules at both fw_unload(D0->D3) and snd_soc_sst_haswell_pcm module removing. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1557e37..bd96629 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -920,6 +920,21 @@ err: return -ENODEV; } +static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) +{ + struct sst_hsw *hsw = pdata->hsw; + struct hsw_pcm_data *pcm_data; + int i; + + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + sst_hsw_runtime_module_free(pcm_data->runtime); + } + if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { + sst_hsw_runtime_module_free(pdata->runtime_waves); + } +} + static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; -- cgit v0.10.2 From 6e5132f79a2e441bde4818abdc813859c8064901 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Sat, 30 May 2015 22:33:57 +0800 Subject: ASoC: intel: Revert "ASoC: Intel: fix broadwell module removing failed issue" This reverts commit 01f202c7b4b40025f3ea4721c52e7f78545e3b07. We shouldn't leave the device as suspended state after module freed, it is not good to do runtime suspend at driver free, here revert this fixing, and replace it with the procedure: suspends firmware ==> frees runtime modules ==> unloads firmware. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index bd96629..23ae040 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1118,10 +1118,8 @@ static int hsw_pcm_remove(struct snd_soc_platform *platform) snd_soc_platform_get_drvdata(platform); int i; - /* execute a suspend call to unload all FW resources */ - if (!pm_runtime_status_suspended(platform->dev)) - pm_runtime_put_sync_suspend(platform->dev); pm_runtime_disable(platform->dev); + hsw_pcm_free_modules(priv_data); for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { if (hsw_dais[i].playback.channels_min) -- cgit v0.10.2 From 2dbc80caf7e93c3d49787cf939fc416873125c1b Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Sat, 30 May 2015 22:33:58 +0800 Subject: ASoC: Intel: check and clear runtime module pointer Add check runtime module pointers before freeing them, and clear them to NULL after freed. With this implemented, we can avoid NULL pointer dereference or double free errors. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 23ae040..f97fa5a 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -928,10 +928,15 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - sst_hsw_runtime_module_free(pcm_data->runtime); + if (pcm_data->runtime){ + sst_hsw_runtime_module_free(pcm_data->runtime); + pcm_data->runtime = NULL; + } } - if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { + if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES) && + pdata->runtime_waves) { sst_hsw_runtime_module_free(pdata->runtime_waves); + pdata->runtime_waves = NULL; } } -- cgit v0.10.2 From edd8ed496b98f1b9d9fda5170a90fe41e7f86e6e Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Sat, 30 May 2015 22:33:59 +0800 Subject: ASoC: Intel: handle haswell pcm suspend including runtime modules freeing It needs free pcm runtime modules before unloading firmware, here add hsw_pcm_suspend() to handle this procedure: suspends firmware ==> frees runtime modules ==> unloads firmware. This fixes the broadwell module unload failed issue. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index f97fa5a..e593e7a 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1209,6 +1209,20 @@ static int hsw_pcm_runtime_idle(struct device *dev) return 0; } +static int hsw_pcm_suspend(struct device *dev) +{ + struct hsw_priv_data *pdata = dev_get_drvdata(dev); + struct sst_hsw *hsw = pdata->hsw; + + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* free all runtime modules */ + hsw_pcm_free_modules(pdata); + /* put the DSP to sleep, fw unloaded after runtime modules freed */ + sst_hsw_dsp_runtime_sleep(hsw); + return 0; +} + static int hsw_pcm_runtime_suspend(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1225,8 +1239,7 @@ static int hsw_pcm_runtime_suspend(struct device *dev) return ret; sst_hsw_set_module_enabled_rtd3(hsw, SST_HSW_MODULE_WAVES); } - sst_hsw_dsp_runtime_suspend(hsw); - sst_hsw_dsp_runtime_sleep(hsw); + hsw_pcm_suspend(dev); pdata->pm_state = HSW_PM_STATE_RTD3; return 0; @@ -1366,10 +1379,7 @@ static int hsw_pcm_prepare(struct device *dev) if (err < 0) dev_err(dev, "failed to save context for PCM %d\n", i); } - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); + hsw_pcm_suspend(dev); } snd_soc_suspend(pdata->soc_card->dev); -- cgit v0.10.2 From bb13f0e08d16a6a303aab786b2aaf2ca76747cfb Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Fri, 29 May 2015 11:56:10 -0700 Subject: ASoC: max98090: read micbias from device property This patch reads max98090 micbias from acpi or dt Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index aa802a2..4e3be66 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -18,6 +18,12 @@ Optional properties: - maxim,dmic-freq: Frequency at which to clock DMIC +- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + Pins on the device (for linking into audio routes): * MIC1 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 3e33ef2..9d80c68 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2422,6 +2422,8 @@ static int max98090_probe(struct snd_soc_codec *codec) struct max98090_cdata *cdata; enum max98090_type devtype; int ret = 0; + int err; + unsigned int micbias; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2506,8 +2508,17 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_MASK); + err = device_property_read_u32(codec->dev, "maxim,micbias", &micbias); + if (err) { + micbias = M98090_MBVSEL_2V8; + dev_info(codec->dev, "use default 2.8v micbias\n"); + } else if (micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) { + dev_err(codec->dev, "micbias out of range 0x%x\n", micbias); + micbias = M98090_MBVSEL_2V8; + } + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, - M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + M98090_MBVSEL_MASK, micbias); max98090_add_widgets(codec); -- cgit v0.10.2 From 859c34bd3cabfc79106f9fcb5c55fb4af3eb3ce2 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 19 May 2015 15:00:40 +0530 Subject: ASoC: Intel: Allocate for the mailbox with max size Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index a7699f3..a12c7bb 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -129,11 +129,31 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc) return -ENOMEM; for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + ipc->msg[i].tx_data = kzalloc(ipc->tx_data_max_size, GFP_KERNEL); + if (ipc->msg[i].tx_data == NULL) + goto free_mem; + + ipc->msg[i].rx_data = kzalloc(ipc->rx_data_max_size, GFP_KERNEL); + if (ipc->msg[i].rx_data == NULL) { + kfree(ipc->msg[i].tx_data); + goto free_mem; + } + init_waitqueue_head(&ipc->msg[i].waitq); list_add(&ipc->msg[i].list, &ipc->empty_list); } return 0; + +free_mem: + while (i > 0) { + kfree(ipc->msg[i-1].tx_data); + kfree(ipc->msg[i-1].rx_data); + --i; + } + kfree(ipc->msg); + + return -ENOMEM; } static void ipc_tx_msgs(struct kthread_work *work) @@ -279,11 +299,18 @@ EXPORT_SYMBOL_GPL(sst_ipc_init); void sst_ipc_fini(struct sst_generic_ipc *ipc) { + int i; + if (ipc->tx_thread) kthread_stop(ipc->tx_thread); - if (ipc->msg) + if (ipc->msg) { + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + kfree(ipc->msg[i].tx_data); + kfree(ipc->msg[i].rx_data); + } kfree(ipc->msg); + } } EXPORT_SYMBOL_GPL(sst_ipc_fini); diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h index 7139afd..ceb7e46 100644 --- a/sound/soc/intel/common/sst-ipc.h +++ b/sound/soc/intel/common/sst-ipc.h @@ -32,9 +32,9 @@ struct ipc_message { u64 header; /* direction wrt host CPU */ - char tx_data[IPC_MAX_MAILBOX_BYTES]; + char *tx_data; size_t tx_size; - char rx_data[IPC_MAX_MAILBOX_BYTES]; + char *rx_data; size_t rx_size; wait_queue_head_t waitq; -- cgit v0.10.2