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: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter)
  ALSA: hda - Fix ALC680 base model capture
  ASoC: Remove DSP mode support for WM8776
  ALSA: hda - Add quirk for Dell Vostro 1220
  ALSA: riptide - Fix detection / load of firmware files

diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 6a664c3..7dc97d1 100644 (file)
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1707,6 +1707,7 @@ struct snd_emu10k1 {
        unsigned int card_type;                 /* EMU10K1_CARD_* */
        unsigned int ecard_ctrl;                /* ecard control bits */
        unsigned long dma_mask;                 /* PCI DMA mask */
+       unsigned int delay_pcm_irq;             /* in samples */
        int max_cache_pages;                    /* max memory size / PAGE_SIZE */
        struct snd_dma_buffer silent_page;      /* silent page */
        struct snd_dma_buffer ptb_pages;        /* page table pages */

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index a3b2a64..134fc6c 100644 (file)
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -978,6 +978,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
 {
        if (substream->runtime->trigger_master != substream)
                return 0;
+       /* some drivers might use hw_ptr to recover from the pause -
+          update the hw_ptr now */
+       if (push)
+               snd_pcm_update_hw_ptr(substream);
        /* The jiffies check in snd_pcm_update_hw_ptr*() is done by
         * a delta betwen the current jiffies, this gives a large enough
         * delta, effectively to skip the check once.

diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 4203782..aff8387 100644 (file)
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
 static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
 static int enable_ir[SNDRV_CARDS];
 static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
@@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444);
 MODULE_PARM_DESC(enable_ir, "Enable IR.");
 module_param_array(subsystem, uint, NULL, 0444);
 MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+module_param_array(delay_pcm_irq, uint, NULL, 0444);
+MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0).");
 /*
  * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value  Model:SB0400
  */
@@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
                                      &emu)) < 0)
                goto error;
        card->private_data = emu;
+       emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f;
        if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0)
                goto error;
        if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0)

diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 55b83ef..622bace 100644 (file)
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
                evoice->epcm->ccca_start_addr = start_addr + ccis;
                if (extra) {
                        start_addr += ccis;
-                       end_addr += ccis;
+                       end_addr += ccis + emu->delay_pcm_irq;
                }
                if (stereo && !extra) {
                        snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK);
@@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
        /* Assumption that PT is already 0 so no harm overwriting */
        snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]);
        snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24));
-       snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24));
+       snd_emu10k1_ptr_write(emu, PSST, voice,
+                       (start_addr + (extra ? emu->delay_pcm_irq : 0)) |
+                       (send_amount[2] << 24));
        if (emu->card_capabilities->emu_model)
                pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */
        else 
@@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_
        snd_emu10k1_ptr_write(emu, IP, voice, 0);
 }
 
+static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu,
+               struct snd_emu10k1_pcm *epcm,
+               struct snd_pcm_substream *substream,
+               struct snd_pcm_runtime *runtime)
+{
+       unsigned int ptr, period_pos;
+
+       /* try to sychronize the current position for the interrupt
+          source voice */
+       period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt;
+       period_pos %= runtime->period_size;
+       ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number);
+       ptr &= ~0x00ffffff;
+       ptr |= epcm->ccca_start_addr + period_pos;
+       snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr);
+}
+
 static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
                                        int cmd)
 {
@@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
                /* follow thru */
        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
        case SNDRV_PCM_TRIGGER_RESUME:
+               if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
+                       snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime);
                mix = &emu->pcm_mixer[substream->number];
                snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix);
                snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix);
@@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *
 #endif
        /*
        printk(KERN_DEBUG
-              "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n",
-              ptr, runtime->buffer_size, runtime->period_size);
+              "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n",
+              (long)ptr, (long)runtime->buffer_size,
+              (long)runtime->period_size);
        */
        return ptr;
 }

diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index ffb1ddb..957a311 100644 (file)
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
        if (snd_BUG_ON(!hdr))
                return NULL;
 
+       idx = runtime->period_size >= runtime->buffer_size ?
+                                       (emu->delay_pcm_irq * 2) : 0;
        mutex_lock(&hdr->block_mutex);
-       blk = search_empty(emu, runtime->dma_bytes);
+       blk = search_empty(emu, runtime->dma_bytes + idx);
        if (blk == NULL) {
                mutex_unlock(&hdr->block_mutex);
                return NULL;

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 31b5d9e..c424952 100644 (file)
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
                      CXT5066_DELL_LAPTOP),
        SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
+       SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO),
        SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
        SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
        SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2cd1ae8..a4dd045 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19030,6 +19030,7 @@ static int patch_alc888(struct hda_codec *codec)
 /*
  * ALC680 support
  */
+#define ALC680_DIGIN_NID       ALC880_DIGIN_NID
 #define ALC680_DIGOUT_NID      ALC880_DIGOUT_NID
 #define alc680_modes           alc260_modes
 
@@ -19044,23 +19045,93 @@ static hda_nid_t alc680_adc_nids[3] = {
        0x07, 0x08, 0x09
 };
 
+/*
+ * Analog capture ADC cgange
+ */
+static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+                                     struct hda_codec *codec,
+                                     unsigned int stream_tag,
+                                     unsigned int format,
+                                     struct snd_pcm_substream *substream)
+{
+       struct alc_spec *spec = codec->spec;
+       struct auto_pin_cfg *cfg = &spec->autocfg;
+       unsigned int pre_mic, pre_line;
+
+       pre_mic  = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
+       pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]);
+
+       spec->cur_adc_stream_tag = stream_tag;
+       spec->cur_adc_format = format;
+
+       if (pre_mic || pre_line) {
+               if (pre_mic)
+                       snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0,
+                                                                       format);
+               else
+                       snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0,
+                                                                       format);
+       } else
+               snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format);
+       return 0;
+}
+
+static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+                                     struct hda_codec *codec,
+                                     struct snd_pcm_substream *substream)
+{
+       snd_hda_codec_cleanup_stream(codec, 0x07);
+       snd_hda_codec_cleanup_stream(codec, 0x08);
+       snd_hda_codec_cleanup_stream(codec, 0x09);
+       return 0;
+}
+
+static struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
+       .substreams = 1, /* can be overridden */
+       .channels_min = 2,
+       .channels_max = 2,
+       /* NID is set in alc_build_pcms */
+       .ops = {
+               .prepare = alc680_capture_pcm_prepare,
+               .cleanup = alc680_capture_pcm_cleanup
+       },
+};
+
 static struct snd_kcontrol_new alc680_base_mixer[] = {
        /* output mixer control */
        HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
        HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
        HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
        HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT),
        HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT),
        { }
 };
 
-static struct snd_kcontrol_new alc680_capture_mixer[] = {
-       HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
-       HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
-       HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
-       HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
-       HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
-       HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+static struct hda_bind_ctls alc680_bind_cap_vol = {
+       .ops = &snd_hda_bind_vol,
+       .values = {
+               HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+               HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+               HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+               0
+       },
+};
+
+static struct hda_bind_ctls alc680_bind_cap_switch = {
+       .ops = &snd_hda_bind_sw,
+       .values = {
+               HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+               HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+               HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+               0
+       },
+};
+
+static struct snd_kcontrol_new alc680_master_capture_mixer[] = {
+       HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
+       HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
        { } /* end */
 };
 
@@ -19068,25 +19139,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = {
  * generic initialization of ADC, input mixers and output mixers
  */
 static struct hda_verb alc680_init_verbs[] = {
-       /* Unmute DAC0-1 and set vol = 0 */
-       {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-       {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-       {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+       {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+       {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+       {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 
-       {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-       {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-       {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
-       {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-       {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+       {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+       {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+       {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+       {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+       {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+       {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 
        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
        {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
        {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
        {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+       {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT   | AC_USRSP_EN},
+       {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT  | AC_USRSP_EN},
+
        { }
 };
 
+/* toggle speaker-output according to the hp-jack state */
+static void alc680_base_setup(struct hda_codec *codec)
+{
+       struct alc_spec *spec = codec->spec;
+
+       spec->autocfg.hp_pins[0] = 0x16;
+       spec->autocfg.speaker_pins[0] = 0x14;
+       spec->autocfg.speaker_pins[1] = 0x15;
+       spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18;
+       spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19;
+}
+
+static void alc680_rec_autoswitch(struct hda_codec *codec)
+{
+       struct alc_spec *spec = codec->spec;
+       struct auto_pin_cfg *cfg = &spec->autocfg;
+       unsigned int present;
+       hda_nid_t new_adc;
+
+       present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
+
+       new_adc = present ? 0x8 : 0x7;
+       __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1);
+       snd_hda_codec_setup_stream(codec, new_adc,
+                                  spec->cur_adc_stream_tag, 0,
+                                  spec->cur_adc_format);
+
+}
+
+static void alc680_unsol_event(struct hda_codec *codec,
+                                          unsigned int res)
+{
+       if ((res >> 26) == ALC880_HP_EVENT)
+               alc_automute_amp(codec);
+       if ((res >> 26) == ALC880_MIC_EVENT)
+               alc680_rec_autoswitch(codec);
+}
+
+static void alc680_inithook(struct hda_codec *codec)
+{
+       alc_automute_amp(codec);
+       alc680_rec_autoswitch(codec);
+}
+
 /* create input playback/capture controls for the given pin */
 static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
                                    const char *ctlname, int idx)
@@ -19197,13 +19316,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec)
 #define alc680_pcm_analog_capture      alc880_pcm_analog_capture
 #define alc680_pcm_analog_alt_capture  alc880_pcm_analog_alt_capture
 #define alc680_pcm_digital_playback    alc880_pcm_digital_playback
-
-static struct hda_input_mux alc680_capture_source = {
-       .num_items = 1,
-       .items = {
-               { "Mic", 0x0 },
-       },
-};
+#define alc680_pcm_digital_capture     alc880_pcm_digital_capture
 
 /*
  * BIOS auto configuration
@@ -19218,6 +19331,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
                                           alc680_ignore);
        if (err < 0)
                return err;
+
        if (!spec->autocfg.line_outs) {
                if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
                        spec->multiout.max_channels = 2;
@@ -19239,8 +19353,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
                add_mixer(spec, spec->kctls.list);
 
        add_verb(spec, alc680_init_verbs);
-       spec->num_mux_defs = 1;
-       spec->input_mux = &alc680_capture_source;
 
        err = alc_auto_add_mic_boost(codec);
        if (err < 0)
@@ -19279,17 +19391,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = {
 static struct alc_config_preset alc680_presets[] = {
        [ALC680_BASE] = {
                .mixers = { alc680_base_mixer },
-               .cap_mixer =  alc680_capture_mixer,
+               .cap_mixer =  alc680_master_capture_mixer,
                .init_verbs = { alc680_init_verbs },
                .num_dacs = ARRAY_SIZE(alc680_dac_nids),
                .dac_nids = alc680_dac_nids,
-               .num_adc_nids = ARRAY_SIZE(alc680_adc_nids),
-               .adc_nids = alc680_adc_nids,
-               .hp_nid = 0x04,
                .dig_out_nid = ALC680_DIGOUT_NID,
                .num_channel_mode = ARRAY_SIZE(alc680_modes),
                .channel_mode = alc680_modes,
-               .input_mux = &alc680_capture_source,
+               .unsol_event = alc680_unsol_event,
+               .setup = alc680_base_setup,
+               .init_hook = alc680_inithook,
+
        },
 };
 
@@ -19333,9 +19445,9 @@ static int patch_alc680(struct hda_codec *codec)
                setup_preset(codec, &alc680_presets[board_config]);
 
        spec->stream_analog_playback = &alc680_pcm_analog_playback;
-       spec->stream_analog_capture = &alc680_pcm_analog_capture;
-       spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture;
+       spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
        spec->stream_digital_playback = &alc680_pcm_digital_playback;
+       spec->stream_digital_capture = &alc680_pcm_digital_capture;
 
        if (!spec->adc_nids) {
                spec->adc_nids = alc680_adc_nids;

diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index f64fb7d..ad5202e 100644 (file)
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
                    firmware.firmware.ASIC, firmware.firmware.CODEC,
                    firmware.firmware.AUXDSP, firmware.firmware.PROG);
 
+       if (!chip)
+               return 1;
+
        for (i = 0; i < FIRMWARE_VERSIONS; i++) {
                if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware)))
-                       break;
-       }
-       if (i >= FIRMWARE_VERSIONS)
-               return 0; /* no match */
+                       return 1; /* OK */
 
-       if (!chip)
-               return 1; /* OK */
+       }
 
        snd_printdd("Writing Firmware\n");
        if (!chip->fw_entry) {

diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 4e212ed..f8154e6 100644 (file)
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
        case SND_SOC_DAIFMT_LEFT_J:
                iface |= 0x0001;
                break;
-               /* FIXME: CHECK A/B */
-       case SND_SOC_DAIFMT_DSP_A:
-               iface |= 0x0003;
-               break;
-       case SND_SOC_DAIFMT_DSP_B:
-               iface |= 0x0007;
-               break;
        default:
                return -EINVAL;
        }