Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix unused warnings when !SND_HDA_NEEDS_RESUME
  ALSA: hda - Add a fix-up for Acer dmic with ALC271x codec
  ASoC: add a module alias to the FSI driver
  ALSA: emu10k1 - Fix "Music" controls to "Synth" controls in documents
  ARM: s3c2440: gta02; Register dfbmcs320 device for BT audio interface
  ASoC: codecs: JZ4740: Fix OOPS
  ASoC: Fix output PGA enabling in wm_hubs CODECs
  ASoC: sn95031: decorate function with __devexit_p()
  ASoC: SAMSUNG: Fix the inverted clocks handling for pcm driver
  ASoC: sst_platform: Fix lock acquring
  ASoC: fsi: driver safely remove for against irq
  ASoC: fsi: modify vague PM control on probe
  ASoC: fsi: take care in failing case of dai register
  MAINTAINERS: Update Samsung ASoC maintainer's id
  ASoC: WM8903: HP and Line out PGA/mixer DAPM fixes
  ASoC: Set left channel volume update bits for WM8994
  ASoC: fix config error path
  ASoC: check channel mismatch between cpu_dai and codec_dai
  ASoC: Tegra: Suspend/resume support

diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
index f5639d4..f4b5988 100644 (file)
--- a/Documentation/sound/alsa/SB-Live-mixer.txt
+++ b/Documentation/sound/alsa/SB-Live-mixer.txt
@@ -87,14 +87,14 @@ accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
 The result is forwarded to the ADC capture FIFO (thus to the standard capture
 PCM device).
 
-name='Music Playback Volume',index=0
+name='Synth Playback Volume',index=0
 
 This control is used to attenuate samples for left and right MIDI FX-bus
 accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
 The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
 
-name='Music Capture Volume',index=0
-name='Music Capture Switch',index=0
+name='Synth Capture Volume',index=0
+name='Synth Capture Switch',index=0
 
 These controls are used to attenuate samples for left and right MIDI FX-bus
 accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.

diff --git a/MAINTAINERS b/MAINTAINERS
index 1e2724e..1380312 100644 (file)
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5396,7 +5396,7 @@ F:        drivers/media/video/*7146*
 F:     include/media/*7146*
 
 SAMSUNG AUDIO (ASoC) DRIVERS
-M:     Jassi Brar <jassi.brar@samsung.com>
+M:     Jassi Brar <jassisinghbrar@gmail.com>
 L:     alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:     Supported
 F:     sound/soc/samsung

diff --git a/arch/arm/mach-s3c2440/mach-gta02.c b/arch/arm/mach-s3c2440/mach-gta02.c
index 0db2411..7166620 100644 (file)
--- a/arch/arm/mach-s3c2440/mach-gta02.c
+++ b/arch/arm/mach-s3c2440/mach-gta02.c
@@ -409,6 +409,10 @@ struct platform_device s3c24xx_pwm_device = {
        .num_resources  = 0,
 };
 
+static struct platform_device gta02_dfbmcs320_device = {
+       .name = "dfbmcs320",
+};
+
 static struct i2c_board_info gta02_i2c_devs[] __initdata = {
        {
                I2C_BOARD_INFO("pcf50633", 0x73),
@@ -523,6 +527,7 @@ static struct platform_device *gta02_devices[] __initdata = {
        &s3c_device_iis,
        &samsung_asoc_dma,
        &s3c_device_i2c0,
+       &gta02_dfbmcs320_device,
        &gta02_buttons_device,
        &s3c_device_adc,
        &s3c_device_ts,

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 430f41d..759ade1 100644 (file)
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -937,6 +937,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
@@ -953,6 +954,7 @@ static void restore_shutup_pins(struct hda_codec *codec)
        }
        codec->pins_shutup = 0;
 }
+#endif
 
 static void init_hda_cache(struct hda_cache_rec *cache,
                           unsigned int record_size);
@@ -1329,6 +1331,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
        }
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
@@ -1340,6 +1343,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec)
                        really_cleanup_stream(codec, p);
        }
 }
+#endif
 
 /*
  * amp access functions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 52928d9..d3bd2c1 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14868,6 +14868,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
        alc_write_coef_idx(codec, 0x1e, coef | 0x80);
 }
 
+static void alc271_fixup_dmic(struct hda_codec *codec,
+                             const struct alc_fixup *fix, int action)
+{
+       static struct hda_verb verbs[] = {
+               {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+               {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+               {}
+       };
+       unsigned int cfg;
+
+       if (strcmp(codec->chip_name, "ALC271X"))
+               return;
+       cfg = snd_hda_codec_get_pincfg(codec, 0x12);
+       if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
+               snd_hda_sequence_write(codec, verbs);
+}
+
 enum {
        ALC269_FIXUP_SONY_VAIO,
        ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -14876,6 +14893,7 @@ enum {
        ALC269_FIXUP_ASUS_G73JW,
        ALC269_FIXUP_LENOVO_EAPD,
        ALC275_FIXUP_SONY_HWEQ,
+       ALC271_FIXUP_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -14929,7 +14947,11 @@ static const struct alc_fixup alc269_fixups[] = {
                .v.func = alc269_fixup_hweq,
                .chained = true,
                .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
-       }
+       },
+       [ALC271_FIXUP_DMIC] = {
+               .type = ALC_FIXUP_FUNC,
+               .v.func = alc271_fixup_dmic,
+       },
 };
 
 static struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -14938,6 +14960,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
        SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
        SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+       SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
        SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
        SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
        SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),

diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index f7cd346..f5ccdbf 100644 (file)
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
        snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
                ARRAY_SIZE(jz4740_codec_dapm_routes));
 
-       snd_soc_dapm_new_widgets(codec);
-
        jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
        return 0;

diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index a54d2a5..4d9fb27 100644 (file)
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -927,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = {
                .owner          = THIS_MODULE,
        },
        .probe          = sn95031_device_probe,
-       .remove         = sn95031_device_remove,
+       .remove         = __devexit_p(sn95031_device_remove),
 };
 
 static int __init sn95031_init(void)

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ae1cadf..f52b623 100644 (file)
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re
        case WM8903_REVISION_NUMBER:
        case WM8903_INTERRUPT_STATUS_1:
        case WM8903_WRITE_SEQUENCER_4:
-       case WM8903_POWER_MANAGEMENT_3:
-       case WM8903_POWER_MANAGEMENT_2:
        case WM8903_DC_SERVO_READBACK_1:
        case WM8903_DC_SERVO_READBACK_2:
        case WM8903_DC_SERVO_READBACK_3:
@@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
 SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
                   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
 
-SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
-                  4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
+                  1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
                   0, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0,
                   NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0,
                   NULL, 0),
 
 SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
                   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
                   NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+                  NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
                   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
                   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
                   NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+                  NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
                   NULL, 0),
 
 SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
@@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = {
        { "Left Speaker PGA", NULL, "Left Speaker Mixer" },
        { "Right Speaker PGA", NULL, "Right Speaker Mixer" },
 
-       { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
-       { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
-       { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
-       { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+       { "HPL_ENA", NULL, "Left Headphone Output PGA" },
+       { "HPR_ENA", NULL, "Right Headphone Output PGA" },
+       { "HPL_ENA_DLY", NULL, "HPL_ENA" },
+       { "HPR_ENA_DLY", NULL, "HPR_ENA" },
+       { "LINEOUTL_ENA", NULL, "Left Line Output PGA" },
+       { "LINEOUTR_ENA", NULL, "Right Line Output PGA" },
+       { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" },
+       { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" },
 
        { "HPL_DCS", NULL, "DCS Master" },
        { "HPR_DCS", NULL, "DCS Master" },

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3290333..84e1bd1 100644 (file)
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
        wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
        /* Latch volume updates (right only; we always do left then right). */
+       snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME,
+                           WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
        snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME,
                            WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
+       snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME,
+                           WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
        snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME,
                            WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
+       snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME,
+                           WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
        snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME,
                            WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
+       snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME,
+                           WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
        snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME,
                            WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
+       snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME,
+                           WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
        snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME,
                            WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
+       snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME,
+                           WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
        snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME,
                            WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
+       snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME,
+                           WM8994_DAC1_VU, WM8994_DAC1_VU);
        snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME,
                            WM8994_DAC1_VU, WM8994_DAC1_VU);
+       snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME,
+                           WM8994_DAC2_VU, WM8994_DAC2_VU);
        snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME,
                            WM8994_DAC2_VU, WM8994_DAC2_VU);
 

diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 7b6b3c1..4005e9a 100644 (file)
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 
        { "SPKL", "Input Switch", "MIXINL" },
        { "SPKL", "IN1LP Switch", "IN1LP" },
-       { "SPKL", "Output Switch", "Left Output Mixer" },
+       { "SPKL", "Output Switch", "Left Output PGA" },
        { "SPKL", NULL, "TOCLK" },
 
        { "SPKR", "Input Switch", "MIXINR" },
        { "SPKR", "IN1RP Switch", "IN1RP" },
-       { "SPKR", "Output Switch", "Right Output Mixer" },
+       { "SPKR", "Output Switch", "Right Output PGA" },
        { "SPKR", NULL, "TOCLK" },
 
        { "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
@@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
        { "SPKOUTRP", NULL, "SPKR Driver" },
        { "SPKOUTRN", NULL, "SPKR Driver" },
 
-       { "Left Headphone Mux", "Mixer", "Left Output Mixer" },
-       { "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+       { "Left Headphone Mux", "Mixer", "Left Output PGA" },
+       { "Right Headphone Mux", "Mixer", "Right Output PGA" },
 
        { "Headphone PGA", NULL, "Left Headphone Mux" },
        { "Headphone PGA", NULL, "Right Headphone Mux" },

diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index b2e9198..d567c32 100644 (file)
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = {
 static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
                                        int state)
 {
-       spin_lock(&stream->status_lock);
+       unsigned long flags;
+       spin_lock_irqsave(&stream->status_lock, flags);
        stream->stream_status = state;
-       spin_unlock(&stream->status_lock);
+       spin_unlock_irqrestore(&stream->status_lock, flags);
 }
 
 static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
 {
        int state;
+       unsigned long flags;
 
-       spin_lock(&stream->status_lock);
+       spin_lock_irqsave(&stream->status_lock, flags);
        state = stream->stream_status;
-       spin_unlock(&stream->status_lock);
+       spin_unlock_irqrestore(&stream->status_lock, flags);
        return state;
 }
 

diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 38aac7d..9c7e8b4 100644 (file)
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai,
        ctl = readl(regs + S3C_PCM_CTL);
 
        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-       case SND_SOC_DAIFMT_NB_NF:
-               /* Nothing to do, NB_NF by default */
+       case SND_SOC_DAIFMT_IB_NF:
+               /* Nothing to do, IB_NF by default */
                break;
        default:
                dev_err(pcm->dev, "Unsupported clock inversion!\n");

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 0c9997e..23c0e83 100644 (file)
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev)
        master->fsib.master     = master;
 
        pm_runtime_enable(&pdev->dev);
-       pm_runtime_resume(&pdev->dev);
        dev_set_drvdata(&pdev->dev, master);
 
+       pm_runtime_get_sync(&pdev->dev);
        fsi_soft_all_reset(master);
+       pm_runtime_put_sync(&pdev->dev);
 
        ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
                          id_entry->name, master);
@@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev)
                goto exit_free_irq;
        }
 
-       return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+       ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
+                                   ARRAY_SIZE(fsi_soc_dai));
+       if (ret < 0) {
+               dev_err(&pdev->dev, "cannot snd dai register\n");
+               goto exit_snd_soc;
+       }
+
+       return ret;
 
+exit_snd_soc:
+       snd_soc_unregister_platform(&pdev->dev);
 exit_free_irq:
        free_irq(irq, master);
 exit_iounmap:
@@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev)
 
        master = dev_get_drvdata(&pdev->dev);
 
-       snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
-       snd_soc_unregister_platform(&pdev->dev);
-
+       free_irq(master->irq, master);
        pm_runtime_disable(&pdev->dev);
 
-       free_irq(master->irq, master);
+       snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+       snd_soc_unregister_platform(&pdev->dev);
 
        iounmap(master->base);
        kfree(master);
@@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit);
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b76b74d..d8562ce 100644 (file)
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
                        runtime->hw.rates |= codec_dai_drv->capture.rates;
        }
 
+       ret = -EINVAL;
        snd_pcm_limit_hw_rates(runtime);
        if (!runtime->hw.rates) {
                printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
@@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
                        codec_dai->name, cpu_dai->name);
                goto config_err;
        }
-       if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+       if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+           runtime->hw.channels_min > runtime->hw.channels_max) {
                printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
                                codec_dai->name, cpu_dai->name);
                goto config_err;
@@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = {
        .resume = snd_soc_resume,
        .poweroff = snd_soc_poweroff,
 };
+EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
 
 /* ASoC platform driver */
 static struct platform_driver soc_driver = {

diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 8585957..556a571 100644 (file)
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = {
        .driver = {
                .name = DRV_NAME,
                .owner = THIS_MODULE,
+               .pm = &snd_soc_pm_ops,
        },
        .probe = tegra_snd_harmony_probe,
        .remove = __devexit_p(tegra_snd_harmony_remove),