diff --git a/MAINTAINERS b/MAINTAINERS index cb78ab59610daf13c3a0915f5e8e02ec20a7e8da..a3e0d10e386e010e158afa99a796e401bbdac0f4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3843,10 +3843,13 @@ F: drivers/tty/serial/ucc_uart.c FREESCALE SOC SOUND DRIVERS M: Timur Tabi +M: Nicolin Chen +M: Xiubo Li L: alsa-devel@alsa-project.org (moderated for non-subscribers) L: linuxppc-dev@lists.ozlabs.org S: Maintained F: sound/soc/fsl/fsl* +F: sound/soc/fsl/imx* F: sound/soc/fsl/mpc8610_hpcd.c FREEVXFS FILESYSTEM diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 36badba2dcec8bde02b67229e4bb8bcd1a3ad3fb..99d7d7fecaad09934090eb6575cfb43131f2dfb4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) +#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) +#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview(codec)) + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || + is_valleyview_plus(codec)) { intel_verify_pin_cvt_connect(codec, per_pin); intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); @@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { /* Verify pin:cvt selections to avoid silent audio after S3. * After S3, the audio driver restores pin:cvt selections * but this can happen before gfx is ready and such selection @@ -2330,9 +2333,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) codec->depop_delay = 0; - } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b38ec3c6e575bd2ee219630ad74412ba434f3b3..d71270a3f73f7ec1826397fe45c00e53fa143a1a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec) spec->pll_coef_idx); val = snd_hda_codec_read(codec, spec->pll_nid, 0, AC_VERB_GET_PROC_COEF, 0); + if (val == -1) + return; snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, spec->pll_coef_idx); snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, @@ -2806,6 +2808,8 @@ static void alc286_shutup(struct hda_codec *codec) static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); + if (val == -1) + return; if (power_up) val |= 1 << 11; else @@ -3264,6 +3268,15 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + + /* on some machine, the BIOS will clear the codec gpio data when enter + * suspend, and won't restore the data after resume, so we restore it + * in the driver. + */ + if (spec->gpio_led) + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); + if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); @@ -5311,27 +5324,30 @@ static void alc269_fill_coef(struct hda_codec *codec) if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { val = alc_read_coef_idx(codec, 0x04); /* Power up output pin */ - alc_write_coef_idx(codec, 0x04, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x04, val | (1<<11)); } if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { val = alc_read_coef_idx(codec, 0xd); - if ((val & 0x0c00) >> 10 != 0x1) { + if (val != -1 && (val & 0x0c00) >> 10 != 0x1) { /* Capless ramp up clock control */ alc_write_coef_idx(codec, 0xd, val | (1<<10)); } val = alc_read_coef_idx(codec, 0x17); - if ((val & 0x01c0) >> 6 != 0x4) { + if (val != -1 && (val & 0x01c0) >> 6 != 0x4) { /* Class D power on reset */ alc_write_coef_idx(codec, 0x17, val | (1<<7)); } } val = alc_read_coef_idx(codec, 0xd); /* Class D */ - alc_write_coef_idx(codec, 0xd, val | (1<<14)); + if (val != -1) + alc_write_coef_idx(codec, 0xd, val | (1<<14)); val = alc_read_coef_idx(codec, 0x4); /* HP */ - alc_write_coef_idx(codec, 0x4, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x4, val | (1<<11)); } /* diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index bd41ee4da078c6c7808364db38d344eab4151a75..2c71f16bd6613fc3e07ca63c4a5b1d8f85efa874 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; + wl = snd_pcm_format_width(params_format(params)); + if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", tdm_slots, tdm_width); @@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, channels = tdm_slots; } else { bclk_target = snd_soc_params_to_bclk(params); + tdm_width = wl; } if (chan_limit && chan_limit < channels) { @@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); - wl = snd_pcm_format_width(params_format(params)); - frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width; reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 163ec3855fd4027e01a1067fa6842688f0175f4b..0c8aefab404ca15cabfaaf43a36d1885983476ed 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds = pcm512x_ramp_step_text); static const struct snd_kcontrol_new pcm512x_controls[] = { -SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, +SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, +SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508da34cf3ba4354ba6d54de685a03d112ca4..6a6b2ff7d7d73daf4b6742242af4247e5585ebcc 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -403,7 +403,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div, bool explicit) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); - mcasp->bclk_div = div; + if (explicit) + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div return 0; } +static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div) +{ + return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); +} + static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -738,7 +746,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, "Inaccurate BCLK: %u Hz / %u != %u Hz\n", mcasp->sysclk_freq, div, bclk_freq); } - davinci_mcasp_set_clkdiv(cpu_dai, 1, div); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } ret = mcasp_common_hw_param(mcasp, substream->stream, diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f54a8fc992913ee837ee8c2914599274982bbba2..f3012b645b51fcfdcdbc5c2e60b8f1d5db11724f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface (ESAI) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e7dd03c3695860dd4da8b00fd6bc2ff3c5..a3b29ed84963459baada6cc9d8ab742f696ee44e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -18,7 +18,6 @@ #include "fsl_esai.h" #include "imx-pcm.h" -#include "fsl_utils.h" #define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 #define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ @@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = { .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, .set_fmt = fsl_esai_set_dai_fmt, - .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask, .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 42edc6f4fc4a839d49140710adcf191f19b7e4a4..03d0a166b6359243a12e046aa64264dc647f4cdd 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { }; static struct sst_acpi_mach baytrail_machines[] = { - { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, - { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, {} }; diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 67673a2c0f415d9f120bf375a3f2a9ca722d7c9a..b4ad98c43e5c6d53a1654e75bc4577d82f15b305 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = { .ops = &sst_byt_ops, }; -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) +int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt = pdata->dsp; @@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) sst_byt_drop_all(byt); dev_dbg(byt->dev, "dsp in reset\n"); - return 0; -} -EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq); - -int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) -{ - struct sst_byt *byt = pdata->dsp; - dev_dbg(byt->dev, "free all blocks and unload fw\n"); sst_fw_unload(byt->fw); diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h index 06a4d202689b3780e13d37a55594862735c0d129..8faff6dcf25d0c1328c799cf40b78620aa3c59d9 100644 --- a/sound/soc/intel/sst-baytrail-ipc.h +++ b/sound/soc/intel/sst-baytrail-ipc.h @@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt, int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt); -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 599401c0c6551d59b1166359e0df5a45e74c8f8d..eab1c7d8518782b01e517e562c6dd7f9eb686d5c 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -59,6 +59,9 @@ struct sst_byt_priv_data { /* DAI data */ struct sst_byt_pcm_data pcm[BYT_PCM_COUNT]; + + /* flag indicating is stream context restore needed after suspend */ + bool restore_stream; }; /* this may get called several times by oss emulation */ @@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - schedule_work(&pcm_data->work); + if (pdata->restore_stream == true) + schedule_work(&pcm_data->work); + else + sst_byt_stream_resume(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: sst_byt_stream_resume(byt, pcm_data->stream); @@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_stop(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: + pdata->restore_stream = false; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; @@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = { }; #ifdef CONFIG_PM -static int sst_byt_pcm_dev_suspend_noirq(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - int ret; - - dev_dbg(dev, "suspending noirq\n"); - - /* at this point all streams will be stopped and context saved */ - ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata); - if (ret < 0) { - dev_err(dev, "failed to suspend %d\n", ret); - return ret; - } - - return ret; -} - static int sst_byt_pcm_dev_suspend_late(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev); int ret; dev_dbg(dev, "suspending late\n"); @@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev) return ret; } + priv_data->restore_stream = true; + return ret; } static int sst_byt_pcm_dev_resume_early(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + int ret; dev_dbg(dev, "resume early\n"); /* load fw and boot DSP */ - return sst_byt_dsp_boot(dev, sst_pdata); -} - -static int sst_byt_pcm_dev_resume(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - - dev_dbg(dev, "resume\n"); + ret = sst_byt_dsp_boot(dev, sst_pdata); + if (ret) + return ret; /* wait for FW to finish booting */ return sst_byt_dsp_wait_for_ready(dev, sst_pdata); } static const struct dev_pm_ops sst_byt_pm_ops = { - .suspend_noirq = sst_byt_pcm_dev_suspend_noirq, .suspend_late = sst_byt_pcm_dev_suspend_late, .resume_early = sst_byt_pcm_dev_resume_early, - .resume = sst_byt_pcm_dev_resume, }; #define SST_BYT_PM_OPS (&sst_byt_pm_ops) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0109f6c2334e528e757b0f128fbf2dddbd3f29de..a8e0974330749f86c894ce520ff43d7c4d0960cc 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .startup = pxa_ssp_startup, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352dc2c62c484a1efba6f04c290367b93864..177bd8639ef93e6b97a20e1cd022224e62cf14bf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - int ret = 0; - if (e->reg != SND_SOC_NOPM) - ret = soc_dapm_read(dapm, e->reg, ®_val); - else + if (e->reg != SND_SOC_NOPM) { + int ret = soc_dapm_read(dapm, e->reg, ®_val); + if (ret) + return ret; + } else { reg_val = dapm_kcontrol_get_value(kcontrol); + } val = (reg_val >> e->shift_l) & e->mask; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); @@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);