diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/codecs/rt5645.c | 22 | ||||
-rw-r--r-- | sound/soc/davinci/davinci-mcasp.c | 14 | ||||
-rw-r--r-- | sound/soc/fsl/fsl-asoc-card.c | 3 | ||||
-rw-r--r-- | sound/soc/intel/haswell/sst-haswell-ipc.c | 20 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 2 | ||||
-rw-r--r-- | sound/soc/soc-utils.c | 9 |
6 files changed, 44 insertions, 26 deletions
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4972bf3efa91..268a28bd1df4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -732,14 +732,14 @@ static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = { static const struct snd_kcontrol_new rt5645_dac_l_mix[] = { SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER, RT5645_M_ADCMIX_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER, RT5645_M_DAC1_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5645_dac_r_mix[] = { SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER, RT5645_M_ADCMIX_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER, RT5645_M_DAC1_R_SFT, 1, 1), }; @@ -1381,7 +1381,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140); - mdelay(5); + msleep(40); rt5645->hp_on = true; } else { /* depop parameters */ @@ -2829,13 +2829,12 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } - - snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001); } else { /* jack out */ rt5645->jack_type = 0; + regmap_update_bits(rt5645->regmap, RT5645_HP_VOL, + RT5645_L_MUTE | RT5645_R_MUTE, + RT5645_L_MUTE | RT5645_R_MUTE); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, @@ -2880,8 +2879,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, rt5645->en_button_func = true; regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); - regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, - RT5645_HP_CB_MASK, RT5645_HP_CB_PU); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); } @@ -3205,6 +3202,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, }, + { + .ident = "Google Ultima", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), + }, + }, { } }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index add6bb99661d..7d45d98a861f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; - int active_serializers, numevt, n; + int active_serializers, numevt; u32 reg; /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) @@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * The number of words for numevt need to be in steps of active * serializers. */ - n = numevt % active_serializers; - if (n) - numevt += (active_serializers - n); + numevt = (numevt / active_serializers) * active_serializers; + while (period_words % numevt && numevt > 0) numevt -= active_serializers; if (numevt <= 0) @@ -1299,6 +1298,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .ops = &davinci_mcasp_dai_ops, .symmetric_samplebits = 1, + .symmetric_rates = 1, }, { .name = "davinci-mcasp.1", @@ -1685,7 +1685,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "common"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, @@ -1702,7 +1702,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_rx_irq_handler, @@ -1717,7 +1717,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "tx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_tx_irq_handler, diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed4827e..96f55ae75c71 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); - return -EINVAL; + ret = -EINVAL; + goto asrc_fail; } /* Common settings for corresponding Freescale CPU DAI driver */ diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f6efa9d4acad..b27f25f70730 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -302,6 +302,10 @@ struct sst_hsw { struct sst_hsw_ipc_dx_reply dx; void *dx_context; dma_addr_t dx_context_paddr; + enum sst_hsw_device_id dx_dev; + enum sst_hsw_device_mclk dx_mclk; + enum sst_hsw_device_mode dx_mode; + u32 dx_clock_divider; /* boot */ wait_queue_head_t boot_wait; @@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, trace_ipc_request("set device config", dev); - config.ssp_interface = dev; - config.clock_frequency = mclk; - config.mode = mode; - config.clock_divider = clock_divider; + hsw->dx_dev = config.ssp_interface = dev; + hsw->dx_mclk = config.clock_frequency = mclk; + hsw->dx_mode = config.mode = mode; + hsw->dx_clock_divider = config.clock_divider = clock_divider; if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER) config.channels = 4; else @@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) return -EIO; } - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); + /* Set ADSP SSP port settings - sadly the FW does not store SSP port + settings as part of the PM context. */ + ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk, + hsw->dx_mode, hsw->dx_clock_divider); if (ret < 0) dev_err(dev, "error: SSP re-initialization failed\n"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f4bf21a5539b..ff8bda471b25 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, default: WARN(1, "Unknown event %d\n", event); - return -EINVAL; + ret = -EINVAL; } out: diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 362c69ac1d6c..53dd085d3ee2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec; SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) +/* + * The dummy CODEC is only meant to be used in situations where there is no + * actual hardware. + * + * If there is actual hardware even if it does not have a control bus + * the hardware will still have constraints like supported samplerates, etc. + * which should be modelled. And the data flow graph also should be modelled + * using DAPM. + */ static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", .playback = { |