diff options
Diffstat (limited to 'sound/soc')
26 files changed, 171 insertions, 81 deletions
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7752860f723..4c23381727a 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + return 0; + err_clk_disable: clk_disable_unprepare(i2s->clk); return ret; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e2a25..35e44e463cf 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); @@ -512,7 +502,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -527,7 +517,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | SSC_BF(TCMR_STTDLY, 1) | SSC_BF(TCMR_START, SSC_START_RISING_RF) - | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | SSC_BF(TCMR_CKS, SSC_CKS_DIV); @@ -556,7 +546,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index e5f2fb884bf..30c673cdc12 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -188,8 +188,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", - "Fixed process flow", "High attenuation with de-emphasis", + "Fixed process flow", "Ringing-less low latency FIR", }; diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2cd4fe46310..1d1c7f8a9af 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, RT286_I2S_CTRL1, 0x0018, d_len_code << 3); dev_dbg(codec->dev, "format val = 0x%x\n", val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); - else - snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); return 0; } diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c3f2decd643..1ff726c2924 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2124,6 +2124,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match); static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, + { "10EC5642", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c0fbe188143..918ada9738b 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2083,10 +2083,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2101,10 +2105,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2212,9 +2220,11 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, - 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT, - 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), /* Input Side */ /* micbias */ diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce610f..aa98be32bb6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -483,21 +483,21 @@ static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* setting i2s data format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_DSP_B: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; case SND_SOC_DAIFMT_I2S: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_RIGHT_J: - i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_MODE_RJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRPOL; break; case SND_SOC_DAIFMT_LEFT_J: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; default: @@ -1462,6 +1462,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (ret) return ret; + /* Need 8 clocks before I2C accesses */ + udelay(1); + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); if (ret) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce054b4..dd222b10ce1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 1d1205702d2..9f2dced046d 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, struct ts3a227e *ts3a227e; struct device *dev = &i2c->dev; int ret; + unsigned int acc_reg; ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); if (ts3a227e == NULL) @@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, ADC_COMPLETE_INT_DISABLE); + /* Read jack status because chip might not trigger interrupt at boot. */ + regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + ts3a227e_jack_report(ts3a227e); + return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b42f6e..b115ed815db 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -717,6 +717,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1380d..75b87c5c0f0 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL Mux", "Left", "AIFINL" }, + { "DACL Mux", "Right", "AIFINR" }, + + { "DACR Mux", "Left", "AIFINL" }, + { "DACR Mux", "Right", "AIFINR" }, + + { "DACL", NULL, "DACL Mux" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "DACR Mux" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae71d9..a96eb497a37 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -556,7 +556,7 @@ static struct { { 22050, 2 }, { 24000, 2 }, { 16000, 3 }, - { 11250, 4 }, + { 11025, 4 }, { 12000, 4 }, { 8000, 5 }, }; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3eddb18fefd..5cc457ef889 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -344,23 +344,27 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) struct snd_ac97 *ac97; int ret = 0; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } - snd_soc_codec_set_drvdata(codec, ac97); - ret = wm9705_reset(codec); if (ret) - goto reset_err; + goto err_put_device; + + ret = device_add(&ac97->dev); + if (ret) + goto err_put_device; + + snd_soc_codec_set_drvdata(codec, ac97); return 0; -reset_err: - snd_soc_free_ac97_codec(ac97); +err_put_device: + put_device(&ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e04643d2bb2..9517571e820 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -666,7 +666,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int ret = 0; - wm9712->ac97 = snd_soc_new_ac97_codec(codec); + wm9712->ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(wm9712->ac97)) { ret = PTR_ERR(wm9712->ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); @@ -675,15 +675,19 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ret = wm9712_reset(codec, 0); if (ret < 0) - goto reset_err; + goto err_put_device; + + ret = device_add(&wm9712->ac97->dev); + if (ret) + goto err_put_device; /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; -reset_err: - snd_soc_free_ac97_codec(wm9712->ac97); +err_put_device: + put_device(&wm9712->ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 71b9d5b0734..6ab1122a387 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1225,7 +1225,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - wm9713->ac97 = snd_soc_new_ac97_codec(codec); + wm9713->ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); @@ -1234,7 +1234,11 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); if (ret < 0) - goto reset_err; + goto err_put_device; + + ret = device_add(&wm9713->ac97->dev); + if (ret) + goto err_put_device; /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; @@ -1242,8 +1246,8 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return 0; -reset_err: - snd_soc_free_ac97_codec(wm9713->ac97); +err_put_device: + put_device(&wm9713->ac97->dev); return ret; } diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f4a10..5e793bbb6b0 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d57ff..059496ed9ad 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->irq = platform_get_irq(pdev, 0); - if (!ssi_private->irq) { + if (ssi_private->irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - return -ENXIO; + return ssi_private->irq; } /* Are the RX and the TX clocks locked? */ diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 4caacb05a62..cd146d4fa80 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) if (ret) goto clk_fail; data->card.num_links = 1; + data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_wm8962_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb9240fdc9b..7fe3009b1c4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Decrease the reference count of the device nodes */ -static int asoc_simple_card_unref(struct platform_device *pdev) +static int asoc_simple_card_unref(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); struct snd_soc_dai_link *dai_link; int num_links; @@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return ret; err: - asoc_simple_card_unref(pdev); + asoc_simple_card_unref(&priv->snd_card); return ret; } @@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); - return asoc_simple_card_unref(pdev); + return asoc_simple_card_unref(card); } static const struct of_device_id asoc_simple_of_match[] = { diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index ef2e8b5766a..b3f9489794a 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -706,6 +706,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba struct list_head *block_list) { struct sst_mem_block *block, *tmp; + struct sst_block_allocator ba_tmp = *ba; u32 end = ba->offset + ba->size, block_end; int err; @@ -730,9 +731,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba if (ba->offset >= block->offset && ba->offset < block_end) { /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; @@ -767,10 +768,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba list_move(&block->list, &dsp->used_block_list); list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c4823136..8156cc1accb 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -651,11 +651,11 @@ static void hsw_notification_work(struct work_struct *work) } /* tell DSP that notification has been handled */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); + sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) @@ -1228,6 +1228,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) struct sst_dsp *sst = hsw->dsp; unsigned long flags; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n"); + return 0; + } + /* dont free DSP streams that are not commited */ if (!stream->commited) goto out; @@ -1415,6 +1420,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) u32 header; int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n"); + return 0; + } + + if (stream->commited) { + dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n"); + return 0; + } + trace_ipc_request("stream alloc", stream->host_id); header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); @@ -1519,6 +1534,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n"); + return 0; + } + trace_ipc_request("stream pause", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, @@ -1535,6 +1555,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n"); + return 0; + } + trace_ipc_request("stream resume", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, @@ -1550,6 +1575,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { int ret, tries = 10; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n"); + return 0; + } + /* dont reset streams that are not commited */ if (!stream->commited) return 0; diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 2ac72eb5e75..b3360139c41 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -350,7 +350,7 @@ static struct sst_machines sst_acpi_bytcr[] = { /* Cherryview-based platforms: CherryTrail and Braswell */ static struct sst_machines sst_acpi_chv[] = { - {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin", + {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8b79cafab1e..c7eb9dd67f6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBM_CFS: /* McBSP slave. FS clock as output */ regs->srgr2 |= FSGM; - regs->pcr0 |= FSXM; + regs->pcr0 |= FSXM | FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 13d8507333b..dcc26eda053 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -335,6 +335,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S24_LE), }, .ops = &rockchip_i2s_dai_ops, + .symmetric_rates = 1, }; static const struct snd_soc_component_driver rockchip_i2s_component = { diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 2e10e9a3837..08d7259bbaa 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -48,15 +48,18 @@ static void soc_ac97_device_release(struct device *dev) } /** - * snd_soc_new_ac97_codec - initailise AC97 device - * @codec: audio codec + * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device + * @codec: The CODEC for which to create the AC'97 device * - * Initialises AC97 codec resources for use by ad-hoc devices only. + * Allocated a new snd_ac97 device and intializes it, but does not yet register + * it. The caller is responsible to either call device_add(&ac97->dev) to + * register the device, or to call put_device(&ac97->dev) to free the device. + * + * Returns: A snd_ac97 device or a PTR_ERR in case of an error. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret; ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); if (ac97 == NULL) @@ -73,7 +76,28 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) codec->component.card->snd_card->number, 0, codec->component.name); - ret = device_register(&ac97->dev); + device_initialize(&ac97->dev); + + return ac97; +} +EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); + +/** + * snd_soc_new_ac97_codec - initailise AC97 device + * @codec: audio codec + * + * Initialises AC97 codec resources for use by ad-hoc devices only. + */ +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +{ + struct snd_ac97 *ac97; + int ret; + + ac97 = snd_soc_alloc_ac97_codec(codec); + if (IS_ERR(ac97)) + return ac97; + + ret = device_add(&ac97->dev); if (ret) { put_device(&ac97->dev); return ERR_PTR(ret); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0..025c38fbe3c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); |