Merge tag 'asoc-v3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.17

Nothing too exciting here, a bunch of driver fixes that came along since
the initial pull request but none that really stand our and a warning
fix in the core.

11 files changed, 43 insertions, 58 deletions

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 2f2e91ac690f..4dfab9573a95 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
         else
                 rates = &arizona_48k_bclk_rates[0];
 
+        wl = snd_pcm_format_width(params_format(params));
+
         if (tdm_slots) {
                 arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
                                 tdm_slots, tdm_width);
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
                 channels = tdm_slots;
         } else {
                 bclk_target = snd_soc_params_to_bclk(params);
+                tdm_width = wl;
         }
 
         if (chan_limit && chan_limit < channels) {
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
         arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
                         rates[bclk], rates[bclk] / lrclk);
 
-        wl = snd_pcm_format_width(params_format(params));
-        frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+        frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width;
 
         reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
 
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 163ec3855fd4..0c8aefab404c 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds =
                         pcm512x_ramp_step_text);
 
 static const struct snd_kcontrol_new pcm512x_controls[] = {
-SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2,
                  PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
 SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
                PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
 SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
                PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
-SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
            PCM512x_RQMR_SHIFT, 1, 1),
 
 SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c28508da34cf..6a6b2ff7d7d7 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -403,7 +403,8 @@ out:
         return ret;
 }
 
-static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+                                      int div, bool explicit)
 {
         struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
@@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
                                ACLKXDIV(div - 1), ACLKXDIV_MASK);
                 mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
                                ACLKRDIV(div - 1), ACLKRDIV_MASK);
-                mcasp->bclk_div = div;
+                if (explicit)
+                        mcasp->bclk_div = div;
                 break;
 
         case 2:         /* BCLK/LRCLK ratio */
@@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
         return 0;
 }
 
+static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+                                    int div)
+{
+        return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+}
+
 static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
                                     unsigned int freq, int dir)
 {
@@ -738,7 +746,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
                                  "Inaccurate BCLK: %u Hz / %u != %u Hz\n",
                                  mcasp->sysclk_freq, div, bclk_freq);
                 }
-                davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
+                __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
         }
 
         ret = mcasp_common_hw_param(mcasp, substream->stream,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f54a8fc99291..f3012b645b51 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI
         tristate "Enhanced Serial Audio Interface (ESAI) module support"
         select REGMAP_MMIO
         select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
-        select SND_SOC_FSL_UTILS
         help
           Say Y if you want to add Enhanced Synchronous Audio Interface
           (ESAI) support for the Freescale CPUs.
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e7dd03..a3b29ed84963 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -18,7 +18,6 @@
 
 #include "fsl_esai.h"
 #include "imx-pcm.h"
-#include "fsl_utils.h"
 
 #define FSL_ESAI_RATES          SNDRV_PCM_RATE_8000_192000
 #define FSL_ESAI_FORMATS        (SNDRV_PCM_FMTBIT_S8 | \
@@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = {
         .hw_params = fsl_esai_hw_params,
         .set_sysclk = fsl_esai_set_dai_sysclk,
         .set_fmt = fsl_esai_set_dai_fmt,
-        .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
         .set_tdm_slot = fsl_esai_set_dai_tdm_slot,
 };
 
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
index 42edc6f4fc4a..03d0a166b635 100644
--- a/sound/soc/intel/sst-acpi.c
+++ b/sound/soc/intel/sst-acpi.c
@@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
 };
 
 static struct sst_acpi_mach baytrail_machines[] = {
-        { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
-        { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" },
+        { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
+        { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
         {}
 };
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index 67673a2c0f41..b4ad98c43e5c 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = {
         .ops = &sst_byt_ops,
 };
 
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
+int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
 {
         struct sst_byt *byt = pdata->dsp;
 
@@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
         sst_byt_drop_all(byt);
         dev_dbg(byt->dev, "dsp in reset\n");
 
-        return 0;
-}
-EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq);
-
-int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
-{
-        struct sst_byt *byt = pdata->dsp;
-
         dev_dbg(byt->dev, "free all blocks and unload fw\n");
         sst_fw_unload(byt->fw);
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
index 06a4d202689b..8faff6dcf25d 100644
--- a/sound/soc/intel/sst-baytrail-ipc.h
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt,
 int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
 void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
 struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 599401c0c655..eab1c7d85187 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -59,6 +59,9 @@ struct sst_byt_priv_data {
 
         /* DAI data */
         struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+
+        /* flag indicating is stream context restore needed after suspend */
+        bool restore_stream;
 };
 
 /* this may get called several times by oss emulation */
@@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
                 sst_byt_stream_start(byt, pcm_data->stream, 0);
                 break;
         case SNDRV_PCM_TRIGGER_RESUME:
-                schedule_work(&pcm_data->work);
+                if (pdata->restore_stream == true)
+                        schedule_work(&pcm_data->work);
+                else
+                        sst_byt_stream_resume(byt, pcm_data->stream);
                 break;
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
                 sst_byt_stream_resume(byt, pcm_data->stream);
@@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
                 sst_byt_stream_stop(byt, pcm_data->stream);
                 break;
         case SNDRV_PCM_TRIGGER_SUSPEND:
+                pdata->restore_stream = false;
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
                 sst_byt_stream_pause(byt, pcm_data->stream);
                 break;
@@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = {
 };
 
 #ifdef CONFIG_PM
-static int sst_byt_pcm_dev_suspend_noirq(struct device *dev)
-{
-        struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-        int ret;
-
-        dev_dbg(dev, "suspending noirq\n");
-
-        /* at this point all streams will be stopped and context saved */
-        ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata);
-        if (ret < 0) {
-                dev_err(dev, "failed to suspend %d\n", ret);
-                return ret;
-        }
-
-        return ret;
-}
-
 static int sst_byt_pcm_dev_suspend_late(struct device *dev)
 {
         struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+        struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev);
         int ret;
 
         dev_dbg(dev, "suspending late\n");
@@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev)
                 return ret;
         }
 
+        priv_data->restore_stream = true;
+
         return ret;
 }
 
 static int sst_byt_pcm_dev_resume_early(struct device *dev)
 {
         struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+        int ret;
 
         dev_dbg(dev, "resume early\n");
 
         /* load fw and boot DSP */
-        return sst_byt_dsp_boot(dev, sst_pdata);
-}
-
-static int sst_byt_pcm_dev_resume(struct device *dev)
-{
-        struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-
-        dev_dbg(dev, "resume\n");
+        ret = sst_byt_dsp_boot(dev, sst_pdata);
+        if (ret)
+                return ret;
 
         /* wait for FW to finish booting */
         return sst_byt_dsp_wait_for_ready(dev, sst_pdata);
 }
 
 static const struct dev_pm_ops sst_byt_pm_ops = {
-        .suspend_noirq = sst_byt_pcm_dev_suspend_noirq,
         .suspend_late = sst_byt_pcm_dev_suspend_late,
         .resume_early = sst_byt_pcm_dev_resume_early,
-        .resume = sst_byt_pcm_dev_resume,
 };
 
 #define SST_BYT_PM_OPS  (&sst_byt_pm_ops)
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 0109f6c2334e..a8e097433074 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
                           SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
                           SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
-                            SNDRV_PCM_FMTBIT_S24_LE |   \
-                            SNDRV_PCM_FMTBIT_S32_LE)
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
         .startup        = pxa_ssp_startup,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8348352dc2c6..177bd8639ef9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
         struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
         struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
         unsigned int reg_val, val;
-        int ret = 0;
 
-        if (e->reg != SND_SOC_NOPM)
-                ret = soc_dapm_read(dapm, e->reg, &reg_val);
-        else
+        if (e->reg != SND_SOC_NOPM) {
+                int ret = soc_dapm_read(dapm, e->reg, &reg_val);
+                if (ret)
+                        return ret;
+        } else {
                 reg_val = dapm_kcontrol_get_value(kcontrol);
+        }
 
         val = (reg_val >> e->shift_l) & e->mask;
         ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
@@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
                 ucontrol->value.enumerated.item[1] = val;
         }
 
-        return ret;
+        return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);