diff options
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/codecs/ab8500-codec.h | 36 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l52.c | 12 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l52.h | 2 | ||||
-rw-r--r-- | sound/soc/codecs/da7213.c | 8 | ||||
-rw-r--r-- | sound/soc/codecs/max98090.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 10 | ||||
-rw-r--r-- | sound/soc/codecs/wm0010.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm5102.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/wm5110.c | 7 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.c | 15 | ||||
-rw-r--r-- | sound/soc/davinci/davinci-mcasp.c | 7 | ||||
-rw-r--r-- | sound/soc/fsl/imx-ssi.c | 6 | ||||
-rw-r--r-- | sound/soc/kirkwood/kirkwood-i2s.c | 5 | ||||
-rw-r--r-- | sound/soc/soc-compress.c | 8 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 49 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 13 |
16 files changed, 103 insertions, 81 deletions
diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 114f69a0c62..306d0bc8455 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -348,25 +348,25 @@ /* AB8500_ADSLOTSELX */ #define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x10 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x20 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x30 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x40 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x50 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x60 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x70 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x80 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0xF0 #define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x01 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x02 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x03 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x04 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x05 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x06 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x07 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x08 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0x0F #define AB8500_ADSLOTSELX_EVEN_SHIFT 0 #define AB8500_ADSLOTSELX_ODD_SHIFT 4 diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0f6f481cec0..987f728718c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = { { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ - { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ @@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -225,7 +227,7 @@ static const char * const mic_bias_level_text[] = { }; static const struct soc_enum mic_bias_level_enum = - SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; @@ -260,7 +262,7 @@ static const char * const hp_gain_num_text[] = { }; static const struct soc_enum hp_gain_enum = - SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5, ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); static const char * const beep_pitch_text[] = { @@ -413,7 +415,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), @@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 6, 0x7f, 0x19, hl_tlv), + 0, 0x7f, 0x19, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 60985c05907..4277012c471 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -157,7 +157,7 @@ #define CS42L52_PB_CTL1_INV_PCMA (1 << 2) #define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) #define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) -#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE_MASK 0x03 #define CS42L52_PB_CTL1_MUTE 3 #define CS42L52_PB_CTL1_UNMUTE 0 diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 41230ad1c3e..4a6f1daf911 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1488,17 +1488,17 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_samplephase) { case DA7213_DMIC_SAMPLE_ON_CLKEDGE: case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_samplephase << DA7213_DMIC_SAMPLEPHASE_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_clk_rate) { case DA7213_DMIC_CLK_3_0MHZ: case DA7213_DMIC_CLK_1_5MHZ: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_clk_rate << DA7213_DMIC_CLK_RATE_SHIFT); break; } diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ce0d36412c9..8d14a76c724 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "irq = %d\n", max98090->irq); ret = request_threaded_irq(max98090->irq, NULL, - max98090_interrupt, IRQF_TRIGGER_FALLING, + max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { dev_err(codec->dev, "request_irq failed: %d\n", diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d60b7c..1514bf845e4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, break; } - - if (found) - snd_soc_dapm_sync(widget->dapm); } - ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); - mutex_unlock(&widget->codec->mutex); + + if (found) + snd_soc_dapm_sync(widget->dapm); + + ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); return ret; } diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8df2b6e1a1a..370af0cbcc9 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -667,6 +667,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) /* On wm0010 only the CLKCTRL1 value is used */ pll_rec.clkctrl1 = wm0010->pll_clkctrl1; + ret = -ENOMEM; len = pll_rec.length + 8; out = kzalloc(len, GFP_KERNEL); if (!out) { diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d3939ee..100fdadda56 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 731884e0477..88ad7db52dd 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5110_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), @@ -976,6 +977,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1eb152cb109..29e95f93d48 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int drc = wm8994_get_drc(kcontrol->id.name); + if (drc < 0) + return drc; ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc]; return 0; @@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + if (block < 0) + return block; + ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; return 0; @@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; int dac; @@ -3831,8 +3836,14 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) ret); } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); - return IRQ_HANDLED; + goto out; } + } else if (!(reg & WM8958_MICD_STS)) { + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + wm8994->mic_detecting = true; + goto out; } if (wm8994->mic_detecting) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc72f2e..81490febac6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 902fab02b85..c6fa03e2114 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -540,11 +540,6 @@ static int imx_ssi_probe(struct platform_device *pdev) clk_prepare_enable(ssi->clk); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - ret = -ENODEV; - goto failed_get_resource; - } - ssi->base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(ssi->base)) { ret = PTR_ERR(ssi->base); @@ -633,7 +628,6 @@ failed_pdev_fiq_alloc: snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); -failed_get_resource: clk_disable_unprepare(ssi->clk); failed_clk: diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index befe68f5928..4c9dad3263c 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -471,11 +471,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "platform_get_resource failed\n"); - return -ENXIO; - } - priv->io = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(priv->io)) return PTR_ERR(priv->io); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7eb3f2..06a8000aa07 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8b..c7051c457b7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai] = 3, + [snd_soc_dapm_dai_in] = 3, + [snd_soc_dapm_dai_out] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -92,7 +93,8 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, - [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_dai_in] = 10, + [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, @@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_out: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) -{ - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) - return w->active; - - return dapm_generic_check_power(w); -} - /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); @@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; - case snd_soc_dapm_dai: - w->power_check = dapm_dai_check_power; - break; default: w->power_check = dapm_always_on_check_power; break; @@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.reg = SND_SOC_NOPM; if (dai->driver->playback.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_in; template.name = dai->driver->playback.stream_name; template.sname = dai->driver->playback.stream_name; @@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } if (dai->driver->capture.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_out; template.name = dai->driver->capture.stream_name; template.sname = dai->driver->capture.stream_name; @@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { - if (dai_w->id != snd_soc_dapm_dai) + switch (dai_w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } dai = dai_w->priv; @@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (w->dapm != dai_w->dapm) continue; - if (w->id == snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: continue; + default: + break; + } if (!w->sname) continue; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8eefa49..ccb6be4d658 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Create any new FE <--> BE connections */ for (i = 0; i < list->num_widgets; i++) { - if (list->widgets[i]->id != snd_soc_dapm_dai) + switch (list->widgets[i]->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, list->widgets[i], stream); @@ -2011,9 +2016,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } |