diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 6a664c3f7c1e426dd0b74129ce9cf81cb07fefa0..7dc97d12253c1bdc494b939e5e856b1b1fa6fc2c 100644 --- 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 a3b2a6479246deca30152bbb7f84e049634f2662..134fc6c2e08dc01eeda84a730545b0532f0588fe 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 4203782d7cb79bec7e7bf47157b32ee04de1efee..aff8387c45cf2e2955d71ee0ca3446808402feaa 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 55b83ef73c630e83b2d098a45cce1f570923df7f..622bace148e3c4e5e4de3efa1936b56245004c69 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 ffb1ddb8dc28ea1d3fbdf574d0c4efbef9772c98..957a311514c8ed9fe34586b518aa463ec279dddd 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 31b5d9eeba68655db32a631160bcf95049b4f1d0..c424952a734e0e48546518fe6b47ccbf959e9aee 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 2cd1ae809e4677a84834654bcdafa3e07ddaf06a..a4dd04524e4391ce7d47f76196b6eeca5e6e9f85 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 f64fb7d988cb57f63876bbc92f0b94d93beb9cfe..ad5202efd7a9e270523cc517e487c6e1997027e3 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 4e212ed62ea609be9e1b3d45a5ac3fb6efea1a38..f8154e661524c64e1098f318d447ba7bfcda3616 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; }