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-rw-r--r--sound/soc/codecs/rt5645.c22
-rw-r--r--sound/soc/davinci/davinci-mcasp.c14
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c3
-rw-r--r--sound/soc/fsl/fsl_ssi.c5
-rw-r--r--sound/soc/intel/haswell/sst-haswell-ipc.c20
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/soc-utils.c9
7 files changed, 47 insertions, 28 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/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 8ec6fb208ea0..37c5cd4d0e59 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
{
- return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
+ return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+ SND_SOC_DAIFMT_AC97;
}
static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
@@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
CCSR_SSI_SCR_TCH_EN);
}
- if (fmt & SND_SOC_DAIFMT_AC97)
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
fsl_ssi_setup_ac97(ssi_private);
return 0;
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 = {
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-30 17:13:59 +0100