Merge branch 'for-linus' into topic/hda-cleanup

Syncing the HD-audio updates for further cleanup works.

diff --git a/MAINTAINERS b/MAINTAINERS
index aefa94841ff3eee6cc124221ab4bcb56dd7d4ca3..8a17e7cf944608963006243e7cb50ac86fed3bf5 100644 (file)
--- 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_realtek.c b/sound/pci/hda/patch_realtek.c
index 6b38ec3c6e575bd2ee219630ad74412ba434f3b3..d71270a3f73f7ec1826397fe45c00e53fa143a1a 100644 (file)
--- 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 bd41ee4da078c6c7808364db38d344eab4151a75..2c71f16bd6613fc3e07ca63c4a5b1d8f85efa874 100644 (file)
--- 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 163ec3855fd4027e01a1067fa6842688f0175f4b..0c8aefab404ca15cabfaaf43a36d1885983476ed 100644 (file)
--- 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 c28508da34cf3ba4354ba6d54de685a03d112ca4..6a6b2ff7d7d73daf4b6742242af4247e5585ebcc 100644 (file)
--- 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 f54a8fc992913ee837ee8c2914599274982bbba2..f3012b645b51fcfdcdbc5c2e60b8f1d5db11724f 100644 (file)
--- 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 72d154e7dd03c3695860dd4da8b00fd6bc2ff3c5..a3b29ed84963459baada6cc9d8ab742f696ee44e 100644 (file)
--- 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 42edc6f4fc4a839d49140710adcf191f19b7e4a4..03d0a166b6359243a12e046aa64264dc647f4cdd 100644 (file)
--- 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 67673a2c0f415d9f120bf375a3f2a9ca722d7c9a..b4ad98c43e5c6d53a1654e75bc4577d82f15b305 100644 (file)
--- 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 06a4d202689b3780e13d37a55594862735c0d129..8faff6dcf25d0c1328c799cf40b78620aa3c59d9 100644 (file)
--- 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 599401c0c6551d59b1166359e0df5a45e74c8f8d..eab1c7d8518782b01e517e562c6dd7f9eb686d5c 100644 (file)
--- 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 0109f6c2334e528e757b0f128fbf2dddbd3f29de..a8e0974330749f86c894ce520ff43d7c4d0960cc 100644 (file)
--- 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 8348352dc2c62c484a1efba6f04c290367b93864..177bd8639ef93e6b97a20e1cd022224e62cf14bf 100644 (file)
--- 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);