Merge branch 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6

Conflicts:
	arch/sh/kernel/process_32.c

Signed-off-by: Paul Mundt <lethal@linux-sh.org>

8 files changed, 187 insertions, 50 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index a3b2a6479246..134fc6c2e08d 100644
--- 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 4203782d7cb7..aff8387c45cf 100644
--- 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 55b83ef73c63..622bace148e3 100644
--- 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 ffb1ddb8dc28..957a311514c8 100644
--- 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 31b5d9eeba68..c424952a734e 100644
--- 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 2cd1ae809e46..a4dd04524e43 100644
--- 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 f64fb7d988cb..ad5202efd7a9 100644
--- 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 4e212ed62ea6..f8154e661524 100644
--- 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;
         }