diff options
Diffstat (limited to 'sound/soc')
36 files changed, 155 insertions, 100 deletions
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index 318c5ba5360f..dfb744381c42 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create); void sport_delete(struct sport_device *sport) { + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); sport_free_resource(sport); + kfree(sport); } EXPORT_SYMBOL(sport_delete); diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c795921c5f6..23b40186f9b8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 8c39dddd7d00..11b1b714b8b5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9cacaf2d52..1cf7a32d1b21 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 6276e352125f..115a40301810 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->mc13xxx; + mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ @@ -657,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_DAC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, @@ -668,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_CODEC, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, @@ -690,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { .id = MC13783_ID_SYNC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8af6a5245b18..df2f99d1d428 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ + {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ @@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; err: diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 982e437799a8..33c0f3d39c87 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 3fd5b29dc933..a3acb7a85f6a 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -702,7 +702,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) } static const struct regmap_config wm2000_regmap = { - .reg_bits = 8, + .reg_bits = 16, .val_bits = 8, .max_register = WM2000_REG_IF_CTL, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..e33d327396ad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -639,6 +625,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -675,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..01ebbcc5c6a4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 0013afe48e66..dc4262eea4b7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = { { 14, 0x0000 }, /* R14 - Power Management 2 */ { 15, 0x0000 }, /* R15 - Power Management 3 */ { 18, 0x0000 }, /* R18 - Power Management 6 */ - { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 20, 0x945E }, /* R20 - Clock Rates 0 */ { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ { 22, 0x0006 }, /* R22 - Clock Rates 2 */ { 24, 0x0050 }, /* R24 - Audio Interface 0 */ diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index eaf65863ec21..ce6720073798 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*250k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + msleep(100); break; case SND_SOC_BIAS_OFF: @@ -3730,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); - regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - - /* VMID back to 2x250k for standby */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK, 0x100); - return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bb62f4b3d563..6c9eeca85b95 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 2; + bclk_rate = params_rate(params) * 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work) int ret; int report; + pm_runtime_get_sync(dev); + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); if (ret < 0) { dev_err(dev, "Failed to read microphone status: %d\n", ret); + pm_runtime_put(dev); return; } @@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); + + pm_runtime_put(dev); } static irqreturn_t wm8994_mic_irq(int irq, void *data) @@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) int reg; bool present; + pm_runtime_get_sync(codec->dev); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); if (reg < 0) { dev_err(codec->dev, "Failed to read jack status: %d\n", reg); mutex_unlock(&wm8994->accdet_lock); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + pm_runtime_get_sync(codec->dev); + /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. */ @@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -4025,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 099e6ec32125..c6d2076a796b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), +SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), @@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, @@ -619,6 +634,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 3eb19fb71d17..d0b8a3287a85 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (wm9713 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); + codec->control_data = wm9713; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..ce5e5cd254dd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* enable FIFO */ + if (dev->txnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } mcasp_start_tx(dev); } else { - if (dev->rxnumevt) /* enable FIFO */ + if (dev->rxnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } mcasp_start_rx(dev); } } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index fb21b17f17f5..199408ec4261 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..81d7728cf67f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f19bb9..b6fa77678d97 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bfa33b1..b3030718c228 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; u32 scr, stat; int ret; + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + /* mclk should already be set */ if (!saif->mclk && saif->mclk_in_use) { dev_err(cpu_dai->dev, "set mclk first\n"); @@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, return ret; } + /* prepare clk in hw_param, enable in trigger */ + clk_prepare(saif->clk); + if (saif != master_saif) { + /* + * Set an initial clock rate for the saif internal logic to work + * properly. This is important when working in EXTMASTER mode + * that uses the other saif's BITCLK&LRCLK but it still needs a + * basic clock which should be fast enough for the internal + * logic. + */ + clk_enable(saif->clk); + ret = clk_set_rate(saif->clk, 24000000); + clk_disable(saif->clk); + if (ret) + return ret; + + clk_prepare(master_saif->clk); + } + scr = __raw_readl(saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_WORD_LENGTH; diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 009533ab8d18..df65f98211ec 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -59,7 +59,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8a9160..d33c48baaf71 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1046083e90a0..acdd3ef14e08 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-mcbsp"); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5a649da9122a..f0feb06615f8 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-pcm-audio"); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38c10fe..b70964ea448c 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->state &= ~ST_RUNNING; prtd->params->ops->stop(prtd->params->ch); break; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index b7b2a1f91425..89b064650f14 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,7 @@ #include <sound/pcm_params.h> #include <plat/audio.h> -#include <plat/dma.h> +#include <mach/dma.h> #include "dma.h" #include "pcm.h" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f219b2f7ee68..c501af6d8dbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } /* If the driver didn't set I/O up try regmap */ - if (!codec->control_data) + if (!codec->write && dev_get_regmap(codec->dev, NULL)) snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (driver->controls) @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd7c49fafd75..f90139b5f50d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -291,8 +291,11 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); - } else + else + dapm->bias_level = level; + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..0c172938b82a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 97c2cac8e92c..8c7f23729446 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm) continue; buf = &substream->dma_buffer; - if (!buf && !buf->area) + if (!buf || !buf->area) continue; dma_free_writecombine(pcm->card->dev, buf->bytes, diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index d684df294c0c..76cb1b363b71 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { @@ -177,7 +176,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) } alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (alc5632->gpio_hp_det == -ENODEV) + if (alc5632->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; ret = snd_soc_of_parse_card_name(card, "nvidia,model"); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5658bcec1931..8d6900c1ee47 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.dst_addr = dmap->addr; - slave_config.src_maxburst = 0; + slave_config.dst_maxburst = 4; } else { slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.src_addr = dmap->addr; - slave_config.dst_maxburst = 0; + slave_config.src_maxburst = 4; } slave_config.slave_id = dmap->req_sel; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0c5bb33d258e..d4f14e492341 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } else if (np) { pdata->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", 0); - if (pdata->gpio_spkr_en == -ENODEV) + if (pdata->gpio_spkr_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios", 0); - if (pdata->gpio_hp_mute == -ENODEV) + if (pdata->gpio_hp_mute == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (pdata->gpio_hp_det == -ENODEV) + if (pdata->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_int_mic_en = of_get_named_gpio(np, "nvidia,int-mic-en-gpios", 0); - if (pdata->gpio_int_mic_en == -ENODEV) + if (pdata->gpio_int_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_ext_mic_en = of_get_named_gpio(np, "nvidia,ext-mic-en-gpios", 0); - if (pdata->gpio_ext_mic_en == -ENODEV) + if (pdata->gpio_ext_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; } diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 62ac0285bfaf..057e28ef770e 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -21,7 +21,7 @@ #include <linux/mfd/dbx500-prcmu.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index ee14d2dac2f5..eb85113d472a 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -19,7 +19,7 @@ #include <linux/slab.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> @@ -663,7 +663,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data) { - int ret = 0; struct resource *res = NULL; struct i2s_controller *i2s_cont; struct ux500_msp *msp; @@ -685,15 +684,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, if (res == NULL) { dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } - msp->registers = ioremap(res->start, (res->end - res->start + 1)); + msp->registers = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); if (msp->registers == NULL) { dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } msp->msp_state = MSP_STATE_IDLE; @@ -705,7 +703,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_err(&pdev->dev, "%s: ERROR: Failed to allocate I2S-controller!\n", __func__); - goto err_i2s_cont; + return -ENOMEM; } i2s_cont->dev.parent = &pdev->dev; i2s_cont->data = (void *)msp; @@ -716,14 +714,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->i2s_cont = i2s_cont; return 0; - -err_i2s_cont: - iounmap(msp->registers); - -err_res: - devm_kfree(&pdev->dev, msp); - - return ret; } void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, @@ -732,11 +722,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); device_unregister(&msp->i2s_cont->dev); - devm_kfree(&pdev->dev, msp->i2s_cont); - - iounmap(msp->registers); - - devm_kfree(&pdev->dev, msp); } MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 7f71b4a0d4bc..2d9136da9865 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -17,7 +17,7 @@ #include <linux/platform_device.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #define MSP_INPUT_FREQ_APB 48000000 |