14 files changed, 74 insertions, 68 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index cb78ab59610d..a3e0d10e386e 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3843,10 +3843,13 @@ F:	drivers/tty/serial/ucc_uart.c
 
 FREESCALE SOC SOUND DRIVERS
 M:      Timur Tabi <timur@tabi.org>
+M:      Nicolin Chen <nicoleotsuka@gmail.com>
+M:      Xiubo Li <Li.Xiubo@freescale.com>
 L:      alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:      linuxppc-dev@lists.ozlabs.org
 S:      Maintained
 F:      sound/soc/fsl/fsl*
+F:      sound/soc/fsl/imx*
 F:      sound/soc/fsl/mpc8610_hpcd.c
 
 FREEVXFS FILESYSTEM
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 36badba2dcec..99d7d7fecaad 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
 #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec))
 
 #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882)
+#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883)
+#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec))
 
 struct hdmi_spec_per_cvt {
         hda_nid_t cvt_nid;
@@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
                             mux_idx);
 
         /* configure unused pins to choose other converters */
-        if (is_haswell_plus(codec) || is_valleyview(codec))
+        if (is_haswell_plus(codec) || is_valleyview_plus(codec))
                 intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx);
 
         snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid);
@@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
                  *   and this can make HW reset converter selection on a pin.
                  */
                 if (eld->eld_valid && !old_eld_valid && per_pin->setup) {
-                        if (is_haswell_plus(codec) || is_valleyview(codec)) {
+                        if (is_haswell_plus(codec) ||
+                                is_valleyview_plus(codec)) {
                                 intel_verify_pin_cvt_connect(codec, per_pin);
                                 intel_not_share_assigned_cvt(codec, pin_nid,
                                                         per_pin->mux_idx);
@@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
         bool non_pcm;
         int pinctl;
 
-        if (is_haswell_plus(codec) || is_valleyview(codec)) {
+        if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
                 /* Verify pin:cvt selections to avoid silent audio after S3.
                  * After S3, the audio driver restores pin:cvt selections
                  * but this can happen before gfx is ready and such selection
@@ -2330,9 +2333,8 @@ static int patch_generic_hdmi(struct hda_codec *codec)
                 intel_haswell_fixup_enable_dp12(codec);
         }
 
-        if (is_haswell(codec) || is_valleyview(codec)) {
+        if (is_haswell_plus(codec) || is_valleyview_plus(codec))
                 codec->depop_delay = 0;
-        }
 
         if (hdmi_parse_codec(codec) < 0) {
                 codec->spec = NULL;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6b38ec3c6e57..d71270a3f73f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec)
                             spec->pll_coef_idx);
         val = snd_hda_codec_read(codec, spec->pll_nid, 0,
                                  AC_VERB_GET_PROC_COEF, 0);
+        if (val == -1)
+                return;
         snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
                             spec->pll_coef_idx);
         snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
@@ -2806,6 +2808,8 @@ static void alc286_shutup(struct hda_codec *codec)
 static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
 {
         int val = alc_read_coef_idx(codec, 0x04);
+        if (val == -1)
+                return;
         if (power_up)
                 val |= 1 << 11;
         else
@@ -3264,6 +3268,15 @@ static int alc269_resume(struct hda_codec *codec)
         snd_hda_codec_resume_cache(codec);
         alc_inv_dmic_sync(codec, true);
         hda_call_check_power_status(codec, 0x01);
+
+        /* on some machine, the BIOS will clear the codec gpio data when enter
+         * suspend, and won't restore the data after resume, so we restore it
+         * in the driver.
+         */
+        if (spec->gpio_led)
+                snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA,
+                            spec->gpio_led);
+
         if (spec->has_alc5505_dsp)
                 alc5505_dsp_resume(codec);
 
@@ -5311,27 +5324,30 @@ static void alc269_fill_coef(struct hda_codec *codec)
         if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
                 val = alc_read_coef_idx(codec, 0x04);
                 /* Power up output pin */
-                alc_write_coef_idx(codec, 0x04, val | (1<<11));
+                if (val != -1)
+                        alc_write_coef_idx(codec, 0x04, val | (1<<11));
         }
 
         if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
                 val = alc_read_coef_idx(codec, 0xd);
-                if ((val & 0x0c00) >> 10 != 0x1) {
+                if (val != -1 && (val & 0x0c00) >> 10 != 0x1) {
                         /* Capless ramp up clock control */
                         alc_write_coef_idx(codec, 0xd, val | (1<<10));
                 }
                 val = alc_read_coef_idx(codec, 0x17);
-                if ((val & 0x01c0) >> 6 != 0x4) {
+                if (val != -1 && (val & 0x01c0) >> 6 != 0x4) {
                         /* Class D power on reset */
                         alc_write_coef_idx(codec, 0x17, val | (1<<7));
                 }
         }
 
         val = alc_read_coef_idx(codec, 0xd); /* Class D */
-        alc_write_coef_idx(codec, 0xd, val | (1<<14));
+        if (val != -1)
+                alc_write_coef_idx(codec, 0xd, val | (1<<14));
 
         val = alc_read_coef_idx(codec, 0x4); /* HP */
-        alc_write_coef_idx(codec, 0x4, val | (1<<11));
+        if (val != -1)
+                alc_write_coef_idx(codec, 0x4, val | (1<<11));
 }
 
 /*
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index bd41ee4da078..2c71f16bd661 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);