Merge branch 'asoc-4.17' into asoc-4.18 to get adau17x1 changes so

further patches can be applied.

8 files changed, 72 insertions, 26 deletions

diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 80c2a06285bb..12bf24c26818 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -502,7 +502,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
         }
 
         if (adau->sigmadsp) {
-                ret = adau17x1_setup_firmware(adau, params_rate(params));
+                ret = adau17x1_setup_firmware(component, params_rate(params));
                 if (ret < 0)
                         return ret;
         }
@@ -835,26 +835,40 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct adau *adau, unsigned int rate)
+int adau17x1_setup_firmware(struct snd_soc_component *component,
+        unsigned int rate)
 {
         int ret;
-        int dspsr;
+        int dspsr, dsp_run;
+        struct adau *adau = snd_soc_component_get_drvdata(component);
+        struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+
+        snd_soc_dapm_mutex_lock(dapm);
 
         ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr);
         if (ret)
-                return ret;
+                goto err;
+
+        ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run);
+        if (ret)
+                goto err;
 
         regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1);
         regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf);
+        regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0);
 
         ret = sigmadsp_setup(adau->sigmadsp, rate);
         if (ret) {
                 regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0);
-                return ret;
+                goto err;
         }
         regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr);
+        regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run);
 
-        return 0;
+err:
+        snd_soc_dapm_mutex_unlock(dapm);
+
+        return ret;
 }
 EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index a7b1cb770814..e6fe87beec07 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -68,7 +68,8 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct adau *adau, unsigned int rate);
+int adau17x1_setup_firmware(struct snd_soc_component *component,
+        unsigned int rate);
 bool adau17x1_has_dsp(struct adau *adau);
 
 #define ADAU17X1_CLOCK_CONTROL                  0x4000
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 40a700493f4c..da8fd98c7f51 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio,
 
         psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8;
 
+        /* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */
+        if (ratio <= 256) {
+                pm = ratio;
+                fp = 1;
+                goto out;
+        }
+
         /* Set the max fluctuation -- 0.1% of the max devisor */
         savesub = (psr ? 1 : 8)  * 256 * maxfp / 1000;
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0823b08923b5..89df2d9f63d7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -217,6 +217,7 @@ struct fsl_ssi_soc_data {
  * @dai_fmt: DAI configuration this device is currently used with
  * @streams: Mask of current active streams: BIT(TX) and BIT(RX)
  * @i2s_net: I2S and Network mode configurations of SCR register
+ *           (this is the initial settings based on the DAI format)
  * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK
  * @use_dma: DMA is used or FIQ with stream filter
  * @use_dual_fifo: DMA with support for dual FIFO mode
@@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
         }
 
         if (!fsl_ssi_is_ac97(ssi)) {
+                /*
+                 * Keep the ssi->i2s_net intact while having a local variable
+                 * to override settings for special use cases. Otherwise, the
+                 * ssi->i2s_net will lose the settings for regular use cases.
+                 */
+                u8 i2s_net = ssi->i2s_net;
+
                 /* Normal + Network mode to send 16-bit data in 32-bit frames */
                 if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16)
-                        ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET;
+                        i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET;
 
                 /* Use Normal mode to send mono data at 1st slot of 2 slots */
                 if (channels == 1)
-                        ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL;
+                        i2s_net = SSI_SCR_I2S_MODE_NORMAL;
 
                 regmap_update_bits(regs, REG_SSI_SCR,
-                                   SSI_SCR_I2S_NET_MASK, ssi->i2s_net);
+                                   SSI_SCR_I2S_NET_MASK, i2s_net);
         }
 
         /* In synchronous mode, the SSI uses STCCR for capture */
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index ceb105cbd461..addac2a8e52a 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL
           for Baytrail Chromebooks but this option is now deprecated and is
           not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead.
 
+config SND_SST_ATOM_HIFI2_PLATFORM
+        tristate
+        select SND_SOC_COMPRESS
+
 config SND_SST_ATOM_HIFI2_PLATFORM_PCI
-        tristate "PCI HiFi2 (Medfield, Merrifield) Platforms"
+        tristate "PCI HiFi2 (Merrifield) Platforms"
         depends on X86 && PCI
         select SND_SST_IPC_PCI
-        select SND_SOC_COMPRESS
+        select SND_SST_ATOM_HIFI2_PLATFORM
         help
-          If you have a Intel Medfield or Merrifield/Edison platform, then
+          If you have a Intel Merrifield/Edison platform, then
           enable this option by saying Y or m. Distros will typically not
-          enable this option: Medfield devices are not available to
-          developers and while Merrifield/Edison can run a mainline kernel with
-          limited functionality it will require a firmware file which
-          is not in the standard firmware tree
+          enable this option: while Merrifield/Edison can run a mainline
+          kernel with limited functionality it will require a firmware file
+          which is not in the standard firmware tree
 
-config SND_SST_ATOM_HIFI2_PLATFORM
+config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
         tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
+        default ACPI
         depends on X86 && ACPI
         select SND_SST_IPC_ACPI
-        select SND_SOC_COMPRESS
+        select SND_SST_ATOM_HIFI2_PLATFORM
         select SND_SOC_ACPI_INTEL_MATCH
         select IOSF_MBI
         help
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 09db2aec12a3..b2f5d2fa354d 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -281,7 +281,7 @@ static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream,
 static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
                                  unsigned int freq)
 {
-        struct clk *parent_clk;
+        struct clk *parent_clk, *mux;
         char *parent_clk_name;
         int ret = 0;
 
@@ -329,14 +329,21 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
                 return -ENODEV;
         }
 
+        mux = clk_get_parent(dmic->fclk);
+        if (IS_ERR(mux)) {
+                dev_err(dmic->dev, "can't get fck mux parent\n");
+                clk_put(parent_clk);
+                return -ENODEV;
+        }
+
         mutex_lock(&dmic->mutex);
         if (dmic->active) {
                 /* disable clock while reparenting */
                 pm_runtime_put_sync(dmic->dev);
-                ret = clk_set_parent(dmic->fclk, parent_clk);
+                ret = clk_set_parent(mux, parent_clk);
                 pm_runtime_get_sync(dmic->dev);
         } else {
-                ret = clk_set_parent(dmic->fclk, parent_clk);
+                ret = clk_set_parent(mux, parent_clk);
         }
         mutex_unlock(&dmic->mutex);
 
@@ -349,6 +356,7 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
         dmic->fclk_freq = freq;
 
 err_busy:
+        clk_put(mux);
         clk_put(parent_clk);
 
         return ret;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f9ac086d8e05..af04d41a4274 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1573,7 +1573,7 @@ static int rsnd_remove(struct platform_device *pdev)
         return ret;
 }
 
-static int rsnd_suspend(struct device *dev)
+static int __maybe_unused rsnd_suspend(struct device *dev)
 {
         struct rsnd_priv *priv = dev_get_drvdata(dev);
 
@@ -1582,7 +1582,7 @@ static int rsnd_suspend(struct device *dev)
         return 0;
 }
 
-static int rsnd_resume(struct device *dev)
+static int __maybe_unused rsnd_resume(struct device *dev)
 {
         struct rsnd_priv *priv = dev_get_drvdata(dev);
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index fa27d0fca6dc..942c6e5eb4b7 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
                         ec->hdr.name);
 
                 kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL);
-                if (kc[i].name == NULL)
+                if (kc[i].name == NULL) {
+                        kfree(se);
                         goto err_se;
+                }
                 kc[i].private_value = (long)se;
                 kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
                 kc[i].access = ec->hdr.access;
@@ -1442,8 +1444,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create(
                         be->hdr.name, be->hdr.access);
 
                 kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL);
-                if (kc[i].name == NULL)
+                if (kc[i].name == NULL) {
+                        kfree(sbe);
                         goto err;
+                }
                 kc[i].private_value = (long)sbe;
                 kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
                 kc[i].access = be->hdr.access;