diff options
Diffstat (limited to 'sound')
27 files changed, 196 insertions, 86 deletions
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 6fd9391..4fa1dbd 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -133,7 +133,7 @@ static int atmel_abdac_prepare_dma(struct atmel_abdac *dac, period_len = frames_to_bytes(runtime, runtime->period_size); cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, - period_len, DMA_TO_DEVICE); + period_len, DMA_MEM_TO_DEV); if (IS_ERR(cdesc)) { dev_dbg(&dac->pdev->dev, "could not prepare cyclic DMA\n"); return PTR_ERR(cdesc); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 73516f6..61dade6 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -102,7 +102,7 @@ static void atmel_ac97c_dma_capture_period_done(void *arg) static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip, struct snd_pcm_substream *substream, - enum dma_data_direction direction) + enum dma_transfer_direction direction) { struct dma_chan *chan; struct dw_cyclic_desc *cdesc; @@ -118,7 +118,7 @@ static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip, return -EINVAL; } - if (direction == DMA_TO_DEVICE) + if (direction == DMA_MEM_TO_DEV) chan = chip->dma.tx_chan; else chan = chip->dma.rx_chan; @@ -133,7 +133,7 @@ static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip, return PTR_ERR(cdesc); } - if (direction == DMA_TO_DEVICE) { + if (direction == DMA_MEM_TO_DEV) { cdesc->period_callback = atmel_ac97c_dma_playback_period_done; set_bit(DMA_TX_READY, &chip->flags); } else { @@ -393,7 +393,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) if (cpu_is_at32ap7000()) { if (!test_bit(DMA_TX_READY, &chip->flags)) retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_TO_DEVICE); + DMA_MEM_TO_DEV); } else { /* Initialize and start the PDC */ writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR); @@ -484,7 +484,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) if (cpu_is_at32ap7000()) { if (!test_bit(DMA_RX_READY, &chip->flags)) retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_FROM_DEVICE); + DMA_DEV_TO_MEM); } else { /* Initialize and start the PDC */ writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR); diff --git a/sound/core/Kconfig b/sound/core/Kconfig index ad40938..b413ed0 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -12,6 +12,9 @@ config SND_HWDEP config SND_RAWMIDI tristate +config SND_COMPRESS_OFFLOAD + tristate + # To be effective this also requires INPUT - users should say: # select SND_JACK if INPUT=y || INPUT=SND # to avoid having to force INPUT on. @@ -154,16 +157,6 @@ config SND_DYNAMIC_MINORS If you are unsure about this, say N here. -config SND_COMPRESS_OFFLOAD - tristate "ALSA Compressed audio offload support" - default n - help - If you want support for offloading compressed audio and have such - a hardware, then you should say Y here and also to the DSP driver - of your platform. - - If you are unsure about this, say N here. - config SND_SUPPORT_OLD_API bool "Support old ALSA API" default y diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 762bb10..f13ad53 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -268,8 +268,14 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) card->shortname, chip->io, chip->irq); // (4) Alloc components. + err = snd_vortex_mixer(chip); + if (err < 0) { + snd_card_free(card); + return err; + } // ADB pcm. - if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) { + err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_PCM); + if (err < 0) { snd_card_free(card); return err; } @@ -299,11 +305,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return err; } #endif - // snd_ac97_mixer and Vortex mixer. - if ((err = snd_vortex_mixer(chip)) < 0) { - snd_card_free(card); - return err; - } if ((err = snd_vortex_midi(chip)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 02f6e08..bb93815 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -105,6 +105,7 @@ #define MIX_SPDIF(x) (vortex->mixspdif[x]) #define NR_WTPB 0x20 /* WT channels per each bank. */ +#define NR_PCM 0x10 /* Structs */ typedef struct { diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 0488633..0ef2f97 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -168,6 +168,7 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) runtime->hw = snd_vortex_playback_hw_adb; #ifdef CHIP_AU8830 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + VORTEX_IS_QUAD(vortex) && VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { runtime->hw.channels_max = 4; snd_pcm_hw_constraint_list(runtime, 0, diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 5b68435..501501e 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - switch (res >> 28) { + res >>= 28; + switch (res) { case ALC_MIC_EVENT: alc88x_simple_mic_automute(codec); break; default: - alc_sku_unsol_event(codec, res); + alc_exec_unsol_event(codec, res); break; } } +static void alc880_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 28); +} + static void alc880_uniwill_p53_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - if ((res >> 28) == ALC_DCVOL_EVENT) + res >>= 28; + if (res == ALC_DCVOL_EVENT) alc880_uniwill_p53_dcvol_automute(codec); else - alc_sku_unsol_event(codec, res); + alc_exec_unsol_event(codec, res); } /* @@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_ch_modes, .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc880_unsol_event, .setup = alc880_lg_setup, .init_hook = alc_hp_automute, #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index bdf0ed4..bb364a5 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) alc889A_mb31_automute(codec); } +static void alc882_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 26); +} + /* * configuration and preset */ @@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = { .channel_mode = alc885_mba21_ch_modes, .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), .input_mux = &alc882_capture_source, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mba21_setup, .init_hook = alc_hp_automute, }, @@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mbp3_setup, .init_hook = alc_hp_automute, }, @@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mb5_setup, .init_hook = alc_hp_automute, }, @@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &macmini3_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_macmini3_setup, .init_hook = alc_hp_automute, }, @@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &alc889A_imac91_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_imac91_setup, .init_hook = alc_hp_automute, }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0852e20..95dfb68 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -469,6 +469,7 @@ struct azx { unsigned int irq_pending_warned :1; unsigned int probing :1; /* codec probing phase */ unsigned int snoop:1; + unsigned int align_buffer_size:1; /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; @@ -1690,7 +1691,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates = hinfo->rates; snd_pcm_limit_hw_rates(runtime); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (align_buffer_size) + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and @@ -2498,6 +2499,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), + SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), @@ -2772,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } /* disable buffer size rounding to 128-byte multiples if supported */ + chip->align_buffer_size = align_buffer_size; if (chip->driver_caps & AZX_DCAPS_BUFSIZE) - align_buffer_size = 0; + chip->align_buffer_size = 0; /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8a32a69..a7a5733 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e82acf..c95c8bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -185,7 +185,6 @@ struct alc_spec { unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ - unsigned int use_jack_tbl:1; /* 1 for model=auto */ /* auto-mute control */ int automute_mode; @@ -621,17 +620,10 @@ static void alc_mic_automute(struct hda_codec *codec) alc_mux_select(codec, 0, spec->int_mic_idx, false); } -/* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +/* handle the specified unsol action (ALC_XXX_EVENT) */ +static void alc_exec_unsol_event(struct hda_codec *codec, int action) { - struct alc_spec *spec = codec->spec; - if (codec->vendor_id == 0x10ec0880) - res >>= 28; - else - res >>= 26; - if (spec->use_jack_tbl) - res = snd_hda_jack_get_action(codec, res); - switch (res) { + switch (action) { case ALC_HP_EVENT: alc_hp_automute(codec); break; @@ -645,6 +637,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_jack_report_sync(codec); } +/* unsolicited event for HP jack sensing */ +static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if (codec->vendor_id == 0x10ec0880) + res >>= 28; + else + res >>= 26; + res = snd_hda_jack_get_action(codec, res); + alc_exec_unsol_event(codec, res); +} + /* call init functions of standard auto-mute helpers */ static void alc_inithook(struct hda_codec *codec) { @@ -1883,7 +1886,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = { }; #endif -static int alc_build_controls(struct hda_codec *codec) +static int __alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct snd_kcontrol *kctl = NULL; @@ -2029,11 +2032,16 @@ static int alc_build_controls(struct hda_codec *codec) alc_free_kctls(codec); /* no longer needed */ - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + return 0; +} + +static int alc_build_controls(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err = __alc_build_controls(codec); if (err < 0) return err; - - return 0; + return snd_hda_jack_add_kctls(codec, &spec->autocfg); } @@ -3233,7 +3241,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, int i, err, noutputs; noutputs = cfg->line_outs; - if (spec->multi_ios > 0) + if (spec->multi_ios > 0 && cfg->line_outs < 3) noutputs += spec->multi_ios; for (i = 0; i < noutputs; i++) { @@ -3904,7 +3912,6 @@ static void set_capture_mixer(struct hda_codec *codec) static void alc_auto_init_std(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->use_jack_tbl = 1; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); @@ -4168,6 +4175,8 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; @@ -4297,6 +4306,8 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) @@ -4691,6 +4702,8 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 87e684f..336cfcd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1596,7 +1596,7 @@ static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, "Dell Studio 1557", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, - "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), + "Dell Studio XPS 1645", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413, "Dell Studio 1558", STAC_DELL_M6_DMIC), {} /* terminator */ @@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a, "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, - "Alienware M17x", STAC_ALIENWARE_M17X), + "Alienware M17x R3", STAC_DELL_EQ), {} /* terminator */ }; @@ -4163,13 +4163,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, return 1; } -static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) +static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) { int i; for (i = 0; i < cfg->hp_outs; i++) if (cfg->hp_pins[i] == nid) return 1; /* nid is a HP-Out */ - + for (i = 0; i < cfg->line_outs; i++) + if (cfg->line_out_pins[i] == nid) + return 1; /* nid is a line-Out */ return 0; /* nid is not a HP-Out */ }; @@ -4375,7 +4377,7 @@ static int stac92xx_init(struct hda_codec *codec) continue; } - if (is_nid_hp_pin(cfg, nid)) + if (is_nid_out_jack_pin(cfg, nid)) continue; /* already has an unsol event */ pinctl = snd_hda_codec_read(codec, nid, 0, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 40b181b..9f3b01b 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -95,13 +95,13 @@ module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); module_param(buggy_semaphore, bool, 0444); MODULE_PARM_DESC(buggy_semaphore, "Enable workaround for hardwares with problematic codec semaphores."); -module_param(buggy_irq, bool, 0444); +module_param(buggy_irq, bint, 0444); MODULE_PARM_DESC(buggy_irq, "Enable workaround for buggy interrupts on some motherboards."); module_param(xbox, bool, 0444); MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection."); module_param(spdif_aclink, int, 0444); MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); -module_param(inside_vm, bool, 0444); +module_param(inside_vm, bint, 0444); MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); /* just for backward compatibility */ diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 478303e..63cff90 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -177,6 +177,7 @@ static void wm8776_registers_init(struct oxygen *chip) struct xonar_wm87x6 *data = chip->model_data; wm8776_write(chip, WM8776_RESET, 0); + wm8776_write(chip, WM8776_PHASESWAP, WM8776_PH_MASK); wm8776_write(chip, WM8776_DACCTRL1, WM8776_DZCEN | WM8776_PL_LEFT_LEFT | WM8776_PL_RIGHT_RIGHT); wm8776_write(chip, WM8776_DACMUTE, chip->dac_mute ? WM8776_DMUTE : 0); diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index e908a81..a561040 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -1,13 +1,13 @@ ## -## Au1200/Au1550 PSC + DBDMA +## Au1200/Au1550/Au1300 PSC + DBDMA ## config SND_SOC_AU1XPSC - tristate "SoC Audio for Au1200/Au1250/Au1550" + tristate "SoC Audio for Au12xx/Au13xx/Au1550" depends on MIPS_ALCHEMY help This option enables support for the Programmable Serial Controllers in AC97 and I2S mode, and the Descriptor-Based DMA - Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC. + Controller (DBDMA) as found on the Au12xx/Au13xx/Au1550 SoC. config SND_SOC_AU1XPSC_I2S tristate @@ -51,12 +51,14 @@ config SND_SOC_DB1000 of boards (DB1000/DB1500/DB1100). config SND_SOC_DB1200 - tristate "DB1200 AC97+I2S audio support" + tristate "DB1200/DB1300/DB1550 Audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_WM9712 select SND_SOC_AU1XPSC_I2S select SND_SOC_WM8731 help - Select this option to enable audio (AC97 or I2S) on the - Alchemy/AMD/RMI DB1200 demoboard. + Select this option to enable audio (AC97 and I2S) on the + Alchemy/AMD/RMI/NetLogic Db1200, Db1550 and Db1300 evaluation boards. + If you need Db1300 touchscreen support, you definitely want to say Y. diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 1c62939..30ea513 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -1,5 +1,5 @@ /* - * DB1200 ASoC audio fabric support code. + * DB1200/DB1300/DB1550 ASoC audio fabric support code. * * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com> * @@ -28,6 +28,18 @@ static struct platform_device_id db1200_pids[] = { }, { .name = "db1200-i2s", .driver_data = 1, + }, { + .name = "db1300-ac97", + .driver_data = 2, + }, { + .name = "db1300-i2s", + .driver_data = 3, + }, { + .name = "db1550-ac97", + .driver_data = 4, + }, { + .name = "db1550-i2s", + .driver_data = 5, }, {}, }; @@ -50,6 +62,27 @@ static struct snd_soc_card db1200_ac97_machine = { .num_links = 1, }; +static struct snd_soc_dai_link db1300_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .codec_dai_name = "wm9712-hifi", + .cpu_dai_name = "au1xpsc_ac97.1", + .platform_name = "au1xpsc-pcm.1", + .codec_name = "wm9712-codec.1", +}; + +static struct snd_soc_card db1300_ac97_machine = { + .name = "DB1300_AC97", + .dai_link = &db1300_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_card db1550_ac97_machine = { + .name = "DB1550_AC97", + .dai_link = &db1200_ac97_dai, + .num_links = 1, +}; + /*------------------------- I2S PART ---------------------------*/ static int db1200_i2s_startup(struct snd_pcm_substream *substream) @@ -100,11 +133,47 @@ static struct snd_soc_card db1200_i2s_machine = { .num_links = 1, }; +static struct snd_soc_dai_link db1300_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .codec_dai_name = "wm8731-hifi", + .cpu_dai_name = "au1xpsc_i2s.2", + .platform_name = "au1xpsc-pcm.2", + .codec_name = "wm8731.0-001b", + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1300_i2s_machine = { + .name = "DB1300_I2S", + .dai_link = &db1300_i2s_dai, + .num_links = 1, +}; + +static struct snd_soc_dai_link db1550_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .codec_dai_name = "wm8731-hifi", + .cpu_dai_name = "au1xpsc_i2s.3", + .platform_name = "au1xpsc-pcm.3", + .codec_name = "wm8731.0-001b", + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1550_i2s_machine = { + .name = "DB1550_I2S", + .dai_link = &db1550_i2s_dai, + .num_links = 1, +}; + /*------------------------- COMMON PART ---------------------------*/ static struct snd_soc_card *db1200_cards[] __devinitdata = { &db1200_ac97_machine, &db1200_i2s_machine, + &db1300_ac97_machine, + &db1300_i2s_machine, + &db1550_ac97_machine, + &db1550_i2s_machine, }; static int __devinit db1200_audio_probe(struct platform_device *pdev) @@ -138,5 +207,5 @@ static struct platform_driver db1200_audio_driver = { module_platform_driver(db1200_audio_driver); MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("DB1200 ASoC audio support"); +MODULE_DESCRIPTION("DB1200/DB1300/DB1550 ASoC audio support"); MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d7bd918..f8863eb 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1457,5 +1457,5 @@ static void __exit sgtl5000_exit(void) module_exit(sgtl5000_exit); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); -MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>"); +MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 2b40c93..7c7fd92 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -444,6 +444,12 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Enable the FLL */ snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); + /* Both overestimates */ + if (Fref < 1000000) + msleep(3); + else + msleep(1); + dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); wm8993->fll_fref = Fref; diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 3fc9613..de83904 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -113,9 +113,9 @@ static int ep93xx_pcm_open(struct snd_pcm_substream *substream) rtd->dma_data.name = dma_params->name; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dma_data.direction = DMA_TO_DEVICE; + rtd->dma_data.direction = DMA_MEM_TO_DEV; else - rtd->dma_data.direction = DMA_FROM_DEVICE; + rtd->dma_data.direction = DMA_DEV_TO_MEM; rtd->dma_chan = dma_request_channel(mask, ep93xx_pcm_dma_filter, &rtd->dma_data); diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 1cf2fe8..5780c9b 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -88,11 +88,13 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, iprtd->dma_data.dma_request = dma_params->dma; /* Try to grab a DMA channel */ - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); - if (!iprtd->dma_chan) - return -EINVAL; + if (!iprtd->dma_chan) { + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; + } switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -107,12 +109,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.direction = DMA_TO_DEVICE; + slave_config.direction = DMA_MEM_TO_DEV; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; slave_config.dst_maxburst = dma_params->burstsize; } else { - slave_config.direction = DMA_FROM_DEVICE; + slave_config.direction = DMA_DEV_TO_MEM; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; slave_config.src_maxburst = dma_params->burstsize; @@ -159,7 +161,7 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period_bytes * iprtd->periods, iprtd->period_bytes, substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_TO_DEVICE : DMA_FROM_DEVICE); + DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); if (!iprtd->desc) { dev_err(&chan->dev->device, "cannot prepare slave dma\n"); return -EINVAL; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 0e12f4e..105f42a 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period_bytes * iprtd->periods, iprtd->period_bytes, substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_TO_DEVICE : DMA_FROM_DEVICE); + DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); if (!iprtd->desc) { dev_err(&chan->dev->device, "cannot prepare slave dma\n"); return -EINVAL; diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 45d11dd..946020a 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -366,7 +366,7 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) goto out3; /* enbale ac97 multifunction pin */ - mfp_set_groupg(nuc900_audio->dev, "nuc900-audio"); + mfp_set_groupg(nuc900_audio->dev, NULL); return 0; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 427ae0d..e4ba17c 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -86,7 +86,7 @@ static void dma_enqueue(struct snd_pcm_substream *substream) dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE); dma_info.direction = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_TO_DEVICE : DMA_FROM_DEVICE); + ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); dma_info.fp = audio_buffdone; dma_info.fp_param = substream; dma_info.period = prtd->dma_period; @@ -171,7 +171,7 @@ static int dma_hw_params(struct snd_pcm_substream *substream, dma_info.client = prtd->params->client; dma_info.direction = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_TO_DEVICE : DMA_FROM_DEVICE); + ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); dma_info.width = prtd->params->dma_size; dma_info.fifo = prtd->params->dma_addr; prtd->params->ch = prtd->params->ops->request( diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index f8f6816..0193e59 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -131,7 +131,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info, sg_dma_address(&sg) = buff; desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, - &sg, 1, DMA_TO_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + &sg, 1, DMA_MEM_TO_DEV, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dev, "Failed to allocate a dma descriptor\n"); return -ENOMEM; @@ -181,7 +181,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info, sg_dma_address(&sg) = buff; desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, - &sg, 1, DMA_FROM_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + &sg, 1, DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dev, "Failed to allocate dma descriptor\n"); return -ENOMEM; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3986520..b5ecf6d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -907,6 +907,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (err < 0) printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); } + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&platform->dapm); + platform->probed = 0; list_del(&platform->card_list); module_put(platform->dev->driver->owner); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3ad1f59..1f55ded 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1426,7 +1426,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) dapm->target_bias_level = SND_SOC_BIAS_ON; break; case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec->active) + if (dapm->codec && dapm->codec->active) dapm->target_bias_level = SND_SOC_BIAS_ON; else dapm->target_bias_level = SND_SOC_BIAS_STANDBY; diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 93931de..2155461 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -134,7 +134,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) sg_dma_address(&sg) = buf_dma_addr; desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_TO_DEVICE : DMA_FROM_DEVICE, + DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(&chan->dev->device, "cannot prepare slave dma\n"); |