diff options
Diffstat (limited to 'sound')
32 files changed, 1663 insertions, 485 deletions
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 678933721735..3ff8cc5f487a 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -437,9 +437,11 @@ static int i2sbus_shutdown(struct macio_dev* dev) } static struct macio_driver i2sbus_drv = { - .name = "soundbus-i2s", - .owner = THIS_MODULE, - .match_table = i2sbus_match, + .driver = { + .name = "soundbus-i2s", + .owner = THIS_MODULE, + .of_match_table = i2sbus_match, + }, .probe = i2sbus_probe, .remove = i2sbus_remove, #ifdef CONFIG_PM diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 428121a7e705..10c3a871a12d 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -657,7 +657,7 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) if (sr & AC97C_SR_CAEVT) { struct snd_pcm_runtime *runtime; int offset, next_period, block_size; - dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", casr & AC97C_CSR_UNRUN ? " UNRUN" : "", diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index f74c7372b3d1..1db586af4f9c 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2578,6 +2578,9 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) if (err) return -err; + memset(&prev_ctl, 0, sizeof(prev_ctl)); + prev_ctl.control_type = -1; + for (idx = 0; idx < 2000; idx++) { err = hpi_mixer_get_control_by_index( ss, asihpi->h_mixer, diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index e89991ea3543..3b4413448226 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,11 +941,11 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static long outstream_get_space_available(struct hpi_hostbuffer_status +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { - return status->size_in_bytes - ((long)(status->host_index) - - (long)(status->dSP_index)); + return status->size_in_bytes - (status->host_index - + status->dSP_index); } static void outstream_write(struct hpi_adapter_obj *pao, @@ -954,7 +954,7 @@ static void outstream_write(struct hpi_adapter_obj *pao, struct hpi_hw_obj *phw = pao->priv; struct bus_master_interface *interface = phw->p_interface_buffer; struct hpi_hostbuffer_status *status; - long space_available; + u32 space_available; if (!phw->outstream_host_buffer_size[phm->obj_index]) { /* there is no BBM buffer, write via message */ @@ -1007,7 +1007,7 @@ static void outstream_write(struct hpi_adapter_obj *pao, } space_available = outstream_get_space_available(status); - if (space_available < (long)phm->u.d.u.data.data_size) { + if (space_available < phm->u.d.u.data.data_size) { phr->error = HPI_ERROR_INVALID_DATASIZE; return; } @@ -1018,7 +1018,7 @@ static void outstream_write(struct hpi_adapter_obj *pao, && hpios_locked_mem_valid(&phw->outstream_host_buffers[phm-> obj_index])) { u8 *p_bbm_data; - long l_first_write; + u32 l_first_write; u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data; if (hpios_locked_mem_get_virt_addr(&phw-> @@ -1248,9 +1248,9 @@ static void instream_start(struct hpi_adapter_obj *pao, hw_message(pao, phm, phr); } -static long instream_get_bytes_available(struct hpi_hostbuffer_status *status) +static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status) { - return (long)(status->dSP_index) - (long)(status->host_index); + return status->dSP_index - status->host_index; } static void instream_read(struct hpi_adapter_obj *pao, @@ -1259,9 +1259,9 @@ static void instream_read(struct hpi_adapter_obj *pao, struct hpi_hw_obj *phw = pao->priv; struct bus_master_interface *interface = phw->p_interface_buffer; struct hpi_hostbuffer_status *status; - long data_available; + u32 data_available; u8 *p_bbm_data; - long l_first_read; + u32 l_first_read; u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data; if (!phw->instream_host_buffer_size[phm->obj_index]) { @@ -1272,7 +1272,7 @@ static void instream_read(struct hpi_adapter_obj *pao, status = &interface->instream_host_buffer_status[phm->obj_index]; data_available = instream_get_bytes_available(status); - if (data_available < (long)phm->u.d.u.data.data_size) { + if (data_available < phm->u.d.u.data.data_size) { phr->error = HPI_ERROR_INVALID_DATASIZE; return; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a3d638c8c1fd..05e8995f9aec 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -396,15 +396,18 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } for (n = prev_nid + 1; n <= val; n++) { if (conns >= max_conns) { - snd_printk(KERN_ERR - "Too many connections\n"); + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + conns, nid); return -EINVAL; } conn_list[conns++] = n; } } else { if (conns >= max_conns) { - snd_printk(KERN_ERR "Too many connections\n"); + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + conns, nid); return -EINVAL; } conn_list[conns++] = val; @@ -784,6 +787,9 @@ static int read_pin_defaults(struct hda_codec *codec) pin->nid = nid; pin->cfg = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + pin->ctrl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); } return 0; } @@ -912,15 +918,38 @@ static void restore_pincfgs(struct hda_codec *codec) void snd_hda_shutup_pins(struct hda_codec *codec) { int i; + /* don't shut up pins when unloading the driver; otherwise it breaks + * the default pin setup at the next load of the driver + */ + if (codec->bus->shutdown) + return; for (i = 0; i < codec->init_pins.used; i++) { struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); /* use read here for syncing after issuing each verb */ snd_hda_codec_read(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); } + codec->pins_shutup = 1; } EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); +/* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ +static void restore_shutup_pins(struct hda_codec *codec) +{ + int i; + if (!codec->pins_shutup) + return; + if (codec->bus->shutdown) + return; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_hda_codec_write(codec, pin->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin->ctrl); + } + codec->pins_shutup = 0; +} + static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); @@ -1539,6 +1568,17 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); #endif /* SND_HDA_NEEDS_RESUME */ +static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, + unsigned int ofs) +{ + u32 caps = query_amp_caps(codec, nid, dir); + /* get num steps */ + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (ofs < caps) + caps -= ofs; + return caps; +} + /** * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer * @@ -1553,23 +1593,17 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, u8 chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); unsigned int ofs = get_amp_offset(kcontrol); - u32 caps; - caps = query_amp_caps(codec, nid, dir); - /* num steps */ - caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; - if (!caps) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = get_amp_max_value(codec, nid, dir, ofs); + if (!uinfo->value.integer.max) { printk(KERN_WARNING "hda_codec: " "num_steps = 0 for NID=0x%x (ctl = %s)\n", nid, kcontrol->id.name); return -EINVAL; } - if (ofs < caps) - caps -= ofs; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = chs == 3 ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = caps; return 0; } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); @@ -1594,8 +1628,14 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, unsigned int ofs, unsigned int val) { + unsigned int maxval; + if (val > 0) val += ofs; + /* ofs = 0: raw max value */ + maxval = get_amp_max_value(codec, nid, dir, 0); + if (val > maxval) + val = maxval; return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val); } @@ -2907,6 +2947,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); restore_pincfgs(codec); /* restore all current pin configs */ + restore_shutup_pins(codec); hda_exec_init_verbs(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); @@ -2972,26 +3013,31 @@ struct hda_rate_tbl { unsigned int hda_fmt; }; +/* rate = base * mult / div */ +#define HDA_RATE(base, mult, div) \ + (AC_FMT_BASE_##base##K | (((mult) - 1) << AC_FMT_MULT_SHIFT) | \ + (((div) - 1) << AC_FMT_DIV_SHIFT)) + static struct hda_rate_tbl rate_bits[] = { /* rate in Hz, ALSA rate bitmask, HDA format value */ /* autodetected value used in snd_hda_query_supported_pcm */ - { 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */ - { 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */ - { 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */ - { 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */ - { 32000, SNDRV_PCM_RATE_32000, 0x0a00 }, /* 2/3 x 48 */ - { 44100, SNDRV_PCM_RATE_44100, 0x4000 }, /* 44 */ - { 48000, SNDRV_PCM_RATE_48000, 0x0000 }, /* 48 */ - { 88200, SNDRV_PCM_RATE_88200, 0x4800 }, /* 2 x 44 */ - { 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */ - { 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */ - { 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */ + { 8000, SNDRV_PCM_RATE_8000, HDA_RATE(48, 1, 6) }, + { 11025, SNDRV_PCM_RATE_11025, HDA_RATE(44, 1, 4) }, + { 16000, SNDRV_PCM_RATE_16000, HDA_RATE(48, 1, 3) }, + { 22050, SNDRV_PCM_RATE_22050, HDA_RATE(44, 1, 2) }, + { 32000, SNDRV_PCM_RATE_32000, HDA_RATE(48, 2, 3) }, + { 44100, SNDRV_PCM_RATE_44100, HDA_RATE(44, 1, 1) }, + { 48000, SNDRV_PCM_RATE_48000, HDA_RATE(48, 1, 1) }, + { 88200, SNDRV_PCM_RATE_88200, HDA_RATE(44, 2, 1) }, + { 96000, SNDRV_PCM_RATE_96000, HDA_RATE(48, 2, 1) }, + { 176400, SNDRV_PCM_RATE_176400, HDA_RATE(44, 4, 1) }, + { 192000, SNDRV_PCM_RATE_192000, HDA_RATE(48, 4, 1) }, #define AC_PAR_PCM_RATE_BITS 11 /* up to bits 10, 384kHZ isn't supported properly */ /* not autodetected value */ - { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */ + { 9600, SNDRV_PCM_RATE_KNOT, HDA_RATE(48, 1, 5) }, { 0 } /* terminator */ }; @@ -3010,7 +3056,8 @@ static struct hda_rate_tbl rate_bits[] = { unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, - unsigned int maxbps) + unsigned int maxbps, + unsigned short spdif_ctls) { int i; unsigned int val = 0; @@ -3033,20 +3080,20 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, switch (snd_pcm_format_width(format)) { case 8: - val |= 0x00; + val |= AC_FMT_BITS_8; break; case 16: - val |= 0x10; + val |= AC_FMT_BITS_16; break; case 20: case 24: case 32: if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= 0x40; + val |= AC_FMT_BITS_32; else if (maxbps >= 24) - val |= 0x30; + val |= AC_FMT_BITS_24; else - val |= 0x20; + val |= AC_FMT_BITS_20; break; default: snd_printdd("invalid format width %d\n", @@ -3054,6 +3101,9 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, return 0; } + if (spdif_ctls & AC_DIG1_NONAUDIO) + val |= AC_FMT_TYPE_NON_PCM; + return val; } EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 49e939e7e5cd..46f75bccf0d3 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -224,6 +224,27 @@ enum { /* Input converter SDI select */ #define AC_SDI_SELECT (0xf<<0) +/* stream format id */ +#define AC_FMT_CHAN_SHIFT 0 +#define AC_FMT_CHAN_MASK (0x0f << 0) +#define AC_FMT_BITS_SHIFT 4 +#define AC_FMT_BITS_MASK (7 << 4) +#define AC_FMT_BITS_8 (0 << 4) +#define AC_FMT_BITS_16 (1 << 4) +#define AC_FMT_BITS_20 (2 << 4) +#define AC_FMT_BITS_24 (3 << 4) +#define AC_FMT_BITS_32 (4 << 4) +#define AC_FMT_DIV_SHIFT 8 +#define AC_FMT_DIV_MASK (7 << 8) +#define AC_FMT_MULT_SHIFT 11 +#define AC_FMT_MULT_MASK (7 << 11) +#define AC_FMT_BASE_SHIFT 14 +#define AC_FMT_BASE_48K (0 << 14) +#define AC_FMT_BASE_44K (1 << 14) +#define AC_FMT_TYPE_SHIFT 15 +#define AC_FMT_TYPE_PCM (0 << 15) +#define AC_FMT_TYPE_NON_PCM (1 << 15) + /* Unsolicited response control */ #define AC_UNSOL_TAG (0x3f<<0) #define AC_UNSOL_ENABLED (1<<7) @@ -364,6 +385,9 @@ enum { #define AC_DIG2_CC (0x7f<<0) /* Pin widget control - 8bit */ +#define AC_PINCTL_EPT (0x3<<0) +#define AC_PINCTL_EPT_NATIVE 0 +#define AC_PINCTL_EPT_HBR 3 #define AC_PINCTL_VREFEN (0x7<<0) #define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ #define AC_PINCTL_VREF_50 1 /* 50% */ @@ -821,6 +845,7 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ @@ -897,7 +922,9 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec); /* the struct for codec->pin_configs */ struct hda_pincfg { hda_nid_t nid; - unsigned int cfg; + unsigned char ctrl; /* current pin control value */ + unsigned char pad; /* reserved */ + unsigned int cfg; /* default configuration */ }; unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid); @@ -925,7 +952,8 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid); unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, - unsigned int maxbps); + unsigned int maxbps, + unsigned short spdif_ctls); int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, unsigned int format); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index a1fc83753cc6..bf3ced51e0f8 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -649,7 +649,9 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->addr == caddr) { + if (codec->vendor_id == vendorid && + codec->subsystem_id == subid && + codec->addr == caddr) { *codecp = codec; break; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dc79564fea30..66d420212d9a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1653,7 +1653,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, runtime->format, - hinfo->maxbps); + hinfo->maxbps, + apcm->codec->spdif_ctls); if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", @@ -1913,11 +1914,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return -1; /* this shouldn't happen! */ - if (wallclk <= azx_dev->period_wallclk && + if (wallclk < (azx_dev->period_wallclk * 5) / 4 && pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) /* NG - it's below the first next period boundary */ return bdl_pos_adj[chip->dev_index] ? 0 : -1; - azx_dev->start_wallclk = wallclk; + azx_dev->start_wallclk += wallclk; return 1; /* OK, it's fine */ } @@ -1960,7 +1961,7 @@ static void azx_irq_pending_work(struct work_struct *work) spin_unlock_irq(&chip->reg_lock); if (!pending) return; - cond_resched(); + msleep(1); } } @@ -2288,6 +2289,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), @@ -2296,6 +2299,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index afbe314a5bf3..b697fd2a6f8b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3662,7 +3662,12 @@ static int patch_ad1984(struct hda_codec *codec) codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; + if (codec->subsystem_id == 0x17aa20fb) { + /* Thinpad X300 does not have the ability to do SPDIF, + or attach to docking station to use SPDIF */ + spec->multiout.dig_out_nid = 0; + } else + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bf2cb5da956..df8b19b17308 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -131,6 +131,8 @@ struct conexant_spec { unsigned int dc_enable; unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */ unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ + + unsigned int beep_amp; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -515,6 +517,15 @@ static struct snd_kcontrol_new cxt_capture_mixers[] = { {} }; +#ifdef CONFIG_SND_HDA_INPUT_BEEP +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new cxt_beep_mixer[] = { + HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), + { } /* end */ +}; +#endif + static const char *slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", @@ -580,16 +591,52 @@ static int conexant_build_controls(struct hda_codec *codec) return err; } +#ifdef CONFIG_SND_HDA_INPUT_BEEP + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = cxt_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + } + } +#endif + + return 0; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int conexant_suspend(struct hda_codec *codec, pm_message_t state) +{ + snd_hda_shutup_pins(codec); return 0; } +#endif static struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .suspend = conexant_suspend, +#endif + .reboot_notify = snd_hda_shutup_pins, }; +#ifdef CONFIG_SND_HDA_INPUT_BEEP +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#endif + /* * EAPD control * the private value = nid | (invert << 8) @@ -1130,9 +1177,10 @@ static int patch_cxt5045(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5045_init_verbs; spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes), - spec->channel_mode = cxt5045_modes, + spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes); + spec->channel_mode = cxt5045_modes; + set_beep_amp(spec, 0x16, 0, 1); codec->patch_ops = conexant_patch_ops; @@ -1211,6 +1259,9 @@ static int patch_cxt5045(struct hda_codec *codec) break; } + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; } @@ -1632,6 +1683,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + /* on ideapad there is an aditional speaker (subwoofer) to mute */ + if (spec->ideapad) + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1888,6 +1944,13 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, #endif } +static struct hda_verb cxt5051_ideapad_init_verbs[] = { + /* Subwoofer */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1917,6 +1980,7 @@ enum { CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */ CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */ + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ CXT5051_MODELS }; @@ -1927,6 +1991,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba", + [CXT5051_IDEAPAD] = "ideapad", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1938,6 +2003,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), {} }; @@ -1972,6 +2038,8 @@ static int patch_cxt5051(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, @@ -1989,6 +2057,10 @@ static int patch_cxt5051(struct hda_codec *codec) break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; + /* Thinkpad X301 does not have S/PDIF wired and no ability + to use a docking station. */ + if (codec->subsystem_id == 0x17aa211f) + spec->multiout.dig_out_nid = 0; break; case CXT5051_F700: spec->init_verbs[0] = cxt5051_f700_init_verbs; @@ -1999,8 +2071,16 @@ static int patch_cxt5051(struct hda_codec *codec) spec->mixers[0] = cxt5051_toshiba_mixers; spec->auto_mic = AUTO_MIC_PORTB; break; + case CXT5051_IDEAPAD: + spec->init_verbs[spec->num_init_verbs++] = + cxt5051_ideapad_init_verbs; + spec->ideapad = 1; + break; } + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; } @@ -2616,7 +2696,6 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { .put = cxt5066_mic_boost_mux_enum_put, .private_value = 0x23 | 0x100, }, - HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), {} }; @@ -2977,8 +3056,10 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), {} }; @@ -3014,6 +3095,8 @@ static int patch_cxt5066(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, cxt5066_models, cxt5066_cfg_tbl); switch (board_config) { @@ -3062,7 +3145,6 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->dell_vostro = 1; spec->mic_boost = 3; /* default 30dB gain */ - snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; @@ -3104,6 +3186,9 @@ static int patch_cxt5066(struct hda_codec *codec) break; } + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 86067ee78632..522e0748ee99 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -52,6 +52,10 @@ struct hdmi_spec { */ struct hda_multi_out multiout; unsigned int codec_type; + + /* misc flags */ + /* PD bit indicates only the update, not the current state */ + unsigned int old_pin_detect:1; }; @@ -616,6 +620,9 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, * Unsolicited events */ +static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld); + static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { struct hdmi_spec *spec = codec->spec; @@ -632,6 +639,12 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (index < 0) return; + if (spec->old_pin_detect) { + if (pind) + hdmi_present_sense(codec, tag, &spec->sink_eld[index]); + pind = spec->sink_eld[index].monitor_present; + } + spec->sink_eld[index].monitor_present = pind; spec->sink_eld[index].eld_valid = eldv; @@ -685,11 +698,51 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) * Callbacks */ -static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, +/* HBR should be Non-PCM, 8 channels */ +#define is_hbr_format(format) \ + ((format & AC_FMT_TYPE_NON_PCM) && (format & AC_FMT_CHAN_MASK) == 7) + +static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int format) { + struct hdmi_spec *spec = codec->spec; int tag; int fmt; + int pinctl; + int new_pinctl = 0; + int i; + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (!(snd_hda_query_pin_caps(codec, spec->pin[i]) & AC_PINCAP_HBR)) + continue; + + pinctl = snd_hda_codec_read(codec, spec->pin[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + new_pinctl = pinctl & ~AC_PINCTL_EPT; + if (is_hbr_format(format)) + new_pinctl |= AC_PINCTL_EPT_HBR; + else + new_pinctl |= AC_PINCTL_EPT_NATIVE; + + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %spinctl=0x%x\n", + spec->pin[i], + pinctl == new_pinctl ? "" : "new-", + new_pinctl); + + if (pinctl != new_pinctl) + snd_hda_codec_write(codec, spec->pin[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + new_pinctl); + } + + if (is_hbr_format(format) && !new_pinctl) { + snd_printdd("hdmi_setup_stream: HBR is not supported\n"); + return -EINVAL; + } tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); @@ -709,6 +762,7 @@ static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, if (fmt != format) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); + return 0; } /* diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index b81d23e42ace..5972d5e7d01f 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -66,8 +66,7 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); - return 0; + return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 3c10c0b149f4..a281836fd472 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -202,8 +202,7 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch_89(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); - return 0; + return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, @@ -478,6 +477,7 @@ static int patch_nvhdmi_8ch_89(struct hda_codec *codec) codec->spec = spec; spec->codec_type = HDA_CODEC_NVIDIA_MCP89; + spec->old_pin_detect = 1; if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; @@ -508,6 +508,7 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) spec->multiout.max_channels = 8; spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; spec->codec_type = HDA_CODEC_NVIDIA_MCP7X; + spec->old_pin_detect = 1; codec->patch_ops = nvhdmi_patch_ops_8ch_7x; @@ -528,6 +529,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) spec->multiout.max_channels = 2; spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; spec->codec_type = HDA_CODEC_NVIDIA_MCP7X; + spec->old_pin_detect = 1; codec->patch_ops = nvhdmi_patch_ops_2ch; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17d4548cc353..6ac53f7de549 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -256,6 +256,13 @@ enum { ALC882_MODEL_LAST, }; +/* ALC680 models */ +enum { + ALC680_BASE, + ALC680_AUTO, + ALC680_MODEL_LAST, +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -326,6 +333,12 @@ struct alc_spec { hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ + /* capture setup for dynamic dual-adc switch */ + unsigned int cur_adc_idx; + hda_nid_t cur_adc; + unsigned int cur_adc_stream_tag; + unsigned int cur_adc_format; + /* capture source */ unsigned int num_mux_defs; const struct hda_input_mux *input_mux; @@ -367,6 +380,7 @@ struct alc_spec { /* other flags */ unsigned int no_analog :1; /* digital I/O only */ + unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */ int init_amp; /* for virtual master */ @@ -833,9 +847,13 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { unsigned int pincap; + unsigned int oldval; + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; - if (pincap & AC_PINCAP_VREF_80) + /* if the default pin setup is vref50, we give it priority */ + if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50) val = PIN_VREF80; else if (pincap & AC_PINCAP_VREF_50) val = PIN_VREF50; @@ -1003,6 +1021,29 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, return -1; } +/* switch the current ADC according to the jack state */ +static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int present; + hda_nid_t new_adc; + + present = snd_hda_jack_detect(codec, spec->ext_mic.pin); + if (present) + spec->cur_adc_idx = 1; + else + spec->cur_adc_idx = 0; + new_adc = spec->adc_nids[spec->cur_adc_idx]; + if (spec->cur_adc && spec->cur_adc != new_adc) { + /* stream is running, let's swap the current ADC */ + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + spec->cur_adc = new_adc; + snd_hda_codec_setup_stream(codec, new_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + } +} + static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1017,6 +1058,11 @@ static void alc_mic_automute(struct hda_codec *codec) if (snd_BUG_ON(!spec->adc_nids)) return; + if (spec->dual_adc_switch) { + alc_dual_mic_adc_auto_switch(codec); + return; + } + cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0]; present = snd_hda_jack_detect(codec, spec->ext_mic.pin); @@ -1267,6 +1313,8 @@ static int alc_auto_parse_customize_define(struct hda_codec *codec) unsigned nid = 0; struct alc_spec *spec = codec->spec; + spec->cdefine.enable_pcbeep = 1; /* assume always enabled */ + ass = codec->subsystem_id & 0xffff; if (ass != codec->bus->pci->subsystem_device && (ass & 1)) goto do_sku; @@ -1497,6 +1545,63 @@ static int alc_read_coef_idx(struct hda_codec *codec, return val; } +/* set right pin controls for digital I/O */ +static void alc_auto_init_digital(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + hda_nid_t pin; + + for (i = 0; i < spec->autocfg.dig_outs; i++) { + pin = spec->autocfg.dig_out_pins[i]; + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_OUT); + } + } + pin = spec->autocfg.dig_in_pin; + if (pin) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_IN); +} + +/* parse digital I/Os and set up NIDs in BIOS auto-parse mode */ +static void alc_auto_parse_digital(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i, err; + hda_nid_t dig_nid; + + /* support multiple SPDIFs; the secondary is set up as a slave */ + for (i = 0; i < spec->autocfg.dig_outs; i++) { + err = snd_hda_get_connections(codec, + spec->autocfg.dig_out_pins[i], + &dig_nid, 1); + if (err < 0) + continue; + if (!i) { + spec->multiout.dig_out_nid = dig_nid; + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } else { + spec->multiout.slave_dig_outs = spec->slave_dig_outs; + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) + break; + spec->slave_dig_outs[i - 1] = dig_nid; + } + } + + if (spec->autocfg.dig_in_pin) { + hda_nid_t dig_nid; + err = snd_hda_get_connections(codec, + spec->autocfg.dig_in_pin, + &dig_nid, 1); + if (err > 0) + spec->dig_in_nid = dig_nid; + } +} + /* * ALC888 */ @@ -2547,7 +2652,7 @@ static struct snd_kcontrol_new alc_beep_mixer[] = { static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - struct snd_kcontrol *kctl; + struct snd_kcontrol *kctl = NULL; struct snd_kcontrol_new *knew; int i, j, err; unsigned int u; @@ -2619,16 +2724,18 @@ static int alc_build_controls(struct hda_codec *codec) } /* assign Capture Source enums to NID */ - kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); - if (!kctl) - kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); - for (i = 0; kctl && i < kctl->count; i++) { - hda_nid_t *nids = spec->capsrc_nids; - if (!nids) - nids = spec->adc_nids; - err = snd_hda_add_nid(codec, kctl, i, nids[i]); - if (err < 0) - return err; + if (spec->capsrc_nids || spec->adc_nids) { + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + hda_nid_t *nids = spec->capsrc_nids; + if (!nids) + nids = spec->adc_nids; + err = snd_hda_add_nid(codec, kctl, i, nids[i]); + if (err < 0) + return err; + } } if (spec->cap_mixer) { const char *kname = kctl ? kctl->id.name : NULL; @@ -3603,6 +3710,41 @@ static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +/* analog capture with dynamic dual-adc changes */ +static int dualmic_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; + spec->cur_adc = spec->adc_nids[spec->cur_adc_idx]; + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + return 0; +} + +static int dualmic_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + spec->cur_adc = 0; + return 0; +} + +static struct hda_pcm_stream dualmic_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0, /* fill later */ + .ops = { + .prepare = dualmic_capture_pcm_prepare, + .cleanup = dualmic_capture_pcm_cleanup + }, +}; /* */ @@ -4932,7 +5074,7 @@ static void alc880_auto_init_input_src(struct hda_codec *codec) static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -4963,25 +5105,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - /* check multiple SPDIF-out (for recent codecs) */ - for (i = 0; i < spec->autocfg.dig_outs; i++) { - hda_nid_t dig_nid; - err = snd_hda_get_connections(codec, - spec->autocfg.dig_out_pins[i], - &dig_nid, 1); - if (err < 0) - continue; - if (!i) - spec->multiout.dig_out_nid = dig_nid; - else { - spec->multiout.slave_dig_outs = spec->slave_dig_outs; - if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) - break; - spec->slave_dig_outs[i - 1] = dig_nid; - } - } - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = ALC880_DIGIN_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -5004,6 +5128,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); alc880_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -5041,6 +5166,39 @@ static void fixup_automic_adc(struct hda_codec *codec) spec->auto_mic = 0; /* disable auto-mic to be sure */ } +/* select or unmute the given capsrc route */ +static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap, + int idx) +{ + if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { + snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, + HDA_AMP_MUTE, 0); + } else { + snd_hda_codec_write_cache(codec, cap, 0, + AC_VERB_SET_CONNECT_SEL, idx); + } +} + +/* set the default connection to that pin */ +static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[i] : spec->adc_nids[i]; + int idx; + + idx = get_connection_index(codec, cap, pin); + if (idx < 0) + continue; + select_or_unmute_capsrc(codec, cap, idx); + return i; /* return the found index */ + } + return -1; /* not found */ +} + /* choose the ADC/MUX containing the input pin and initialize the setup */ static void fixup_single_adc(struct hda_codec *codec) { @@ -5057,33 +5215,24 @@ static void fixup_single_adc(struct hda_codec *codec) } if (!pin) return; - - /* set the default connection to that pin */ - for (i = 0; i < spec->num_adc_nids; i++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[i] : spec->adc_nids[i]; - int idx; - - idx = get_connection_index(codec, cap, pin); - if (idx < 0) - continue; + i = init_capsrc_for_pin(codec, pin); + if (i >= 0) { /* use only this ADC */ if (spec->capsrc_nids) spec->capsrc_nids += i; spec->adc_nids += i; spec->num_adc_nids = 1; - /* select or unmute this route */ - if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { - snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, - HDA_AMP_MUTE, 0); - } else { - snd_hda_codec_write_cache(codec, cap, 0, - AC_VERB_SET_CONNECT_SEL, idx); - } - return; } } +/* initialize dual adcs */ +static void fixup_dual_adc_switch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + init_capsrc_for_pin(codec, spec->ext_mic.pin); + init_capsrc_for_pin(codec, spec->int_mic.pin); +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5097,7 +5246,10 @@ static void set_capture_mixer(struct hda_codec *codec) }; if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { int mux = 0; - if (spec->auto_mic) + int num_adcs = spec->num_adc_nids; + if (spec->dual_adc_switch) + fixup_dual_adc_switch(codec); + else if (spec->auto_mic) fixup_automic_adc(codec); else if (spec->input_mux) { if (spec->input_mux->num_items > 1) @@ -5105,7 +5257,9 @@ static void set_capture_mixer(struct hda_codec *codec) else if (spec->input_mux->num_items == 1) fixup_single_adc(codec); } - spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; + if (spec->dual_adc_switch) + num_adcs = 1; + spec->cap_mixer = caps[mux][num_adcs - 1]; } } @@ -5176,8 +5330,25 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, #ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +static struct snd_pci_quirk beep_white_list[] = { + SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), + SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), + {} +}; + +static inline int has_cdefine_beep(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + const struct snd_pci_quirk *q; + q = snd_pci_quirk_lookup(codec->bus->pci, beep_white_list); + if (q) + return q->value; + return spec->cdefine.enable_pcbeep; +} #else #define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#define has_cdefine_beep(codec) 0 #endif /* @@ -6604,6 +6775,7 @@ static void alc260_auto_init(struct hda_codec *codec) alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); alc260_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -6620,6 +6792,29 @@ static struct hda_amp_list alc260_loopbacks[] = { #endif /* + * Pin config fixes + */ +enum { + PINFIX_HP_DC5750, +}; + +static struct alc_pincfg alc260_hp_dc5750_pinfix[] = { + { 0x11, 0x90130110 }, /* speaker */ + { } +}; + +static const struct alc_fixup alc260_fixups[] = { + [PINFIX_HP_DC5750] = { + .pins = alc260_hp_dc5750_pinfix + }, +}; + +static struct snd_pci_quirk alc260_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750), + {} +}; + +/* * ALC260 configurations */ static const char *alc260_models[ALC260_MODEL_LAST] = { @@ -6818,6 +7013,9 @@ static int patch_alc260(struct hda_codec *codec) board_config = ALC260_AUTO; } + if (board_config == ALC260_AUTO) + alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 1); + if (board_config == ALC260_AUTO) { /* automatic parse from the BIOS config */ err = alc260_parse_auto_config(codec); @@ -6863,6 +7061,9 @@ static int patch_alc260(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); + if (board_config == ALC260_AUTO) + alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 0); + spec->vmaster_nid = 0x08; codec->patch_ops = alc_patch_ops; @@ -6948,7 +7149,7 @@ static struct hda_input_mux mb5_capture_source = { .num_items = 3, .items = { { "Mic", 0x1 }, - { "Line", 0x2 }, + { "Line", 0x7 }, { "CD", 0x4 }, }, }; @@ -6983,7 +7184,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, - { "iMic", 0x1 }, + { "Int Mic", 0x1 }, { "Line", 0x2 }, { "CD", 0x4 }, }, @@ -7469,8 +7670,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), @@ -7853,10 +8054,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)}, { } }; @@ -8554,8 +8754,8 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -9476,11 +9676,16 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), + SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91), SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5), /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, * so apparently no perfect solution yet */ @@ -10241,7 +10446,8 @@ static struct alc_config_preset alc882_presets[] = { * Pin config fixes */ enum { - PINFIX_ABIT_AW9D_MAX + PINFIX_ABIT_AW9D_MAX, + PINFIX_PB_M5210, }; static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { @@ -10251,13 +10457,22 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { { } }; +static const struct hda_verb pb_m5210_verbs[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, + {} +}; + static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { .pins = alc882_abit_aw9d_pinfix }, + [PINFIX_PB_M5210] = { + .verbs = pb_m5210_verbs + }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), {} }; @@ -10422,7 +10637,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; static hda_nid_t alc882_ignore[] = { 0x1d, 0 }; - int i, err; + int err; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc882_ignore); @@ -10452,25 +10667,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - /* check multiple SPDIF-out (for recent codecs) */ - for (i = 0; i < spec->autocfg.dig_outs; i++) { - hda_nid_t dig_nid; - err = snd_hda_get_connections(codec, - spec->autocfg.dig_out_pins[i], - &dig_nid, 1); - if (err < 0) - continue; - if (!i) - spec->multiout.dig_out_nid = dig_nid; - else { - spec->multiout.slave_dig_outs = spec->slave_dig_outs; - if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) - break; - spec->slave_dig_outs[i - 1] = dig_nid; - } - } - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = ALC880_DIGIN_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -10500,6 +10697,7 @@ static void alc882_auto_init(struct hda_codec *codec) alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); alc882_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -10558,10 +10756,12 @@ static int patch_alc882(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; + if (has_cdefine_beep(codec)) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } } if (board_config != ALC882_AUTO) @@ -10611,7 +10811,7 @@ static int patch_alc882(struct hda_codec *codec) set_capture_mixer(codec); - if (spec->cdefine.enable_pcbeep) + if (has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); if (board_config == ALC882_AUTO) @@ -12028,12 +12228,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; dig_only: - if (spec->autocfg.dig_outs) { - spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; - spec->dig_out_type = spec->autocfg.dig_out_type[0]; - } - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = ALC262_DIGIN_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -12065,6 +12260,7 @@ static void alc262_auto_init(struct hda_codec *codec) alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); alc262_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -12427,7 +12623,7 @@ static int patch_alc262(struct hda_codec *codec) } } - if (!spec->no_analog) { + if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) { alc_free(codec); @@ -12478,7 +12674,7 @@ static int patch_alc262(struct hda_codec *codec) } if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(codec); - if (!spec->no_analog && spec->cdefine.enable_pcbeep) + if (!spec->no_analog && has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -12998,10 +13194,14 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x1a: /* ALC259/269 only */ + case 0x1b: /* ALC259/269 only */ case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; default: + snd_printd(KERN_WARNING "hda_codec: " + "ignoring pin 0x%x as unknown\n", nid); return 0; } if (spec->multiout.dac_nids[0] != dac && @@ -13052,7 +13252,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) return err; - } else { + } else if (nid) { err = alc268_new_analog_output(spec, nid, "Speaker", 0); if (err < 0) return err; @@ -13201,10 +13401,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) dig_only: /* digital only support output */ - if (spec->autocfg.dig_outs) { - spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - spec->dig_out_type = spec->autocfg.dig_out_type[0]; - } + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -13234,6 +13431,7 @@ static void alc268_auto_init(struct hda_codec *codec) alc268_auto_init_hp_out(codec); alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -14126,6 +14324,36 @@ static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid) } #endif /* CONFIG_SND_HDA_POWER_SAVE */ +static int alc275_setup_dual_adc(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (codec->vendor_id != 0x10ec0275 || !spec->auto_mic) + return 0; + if ((spec->ext_mic.pin >= 0x18 && spec->int_mic.pin <= 0x13) || + (spec->ext_mic.pin <= 0x12 && spec->int_mic.pin >= 0x18)) { + if (spec->ext_mic.pin <= 0x12) { + spec->private_adc_nids[0] = 0x08; + spec->private_adc_nids[1] = 0x11; + spec->private_capsrc_nids[0] = 0x23; + spec->private_capsrc_nids[1] = 0x22; + } else { + spec->private_adc_nids[0] = 0x11; + spec->private_adc_nids[1] = 0x08; + spec->private_capsrc_nids[0] = 0x22; + spec->private_capsrc_nids[1] = 0x23; + } + spec->adc_nids = spec->private_adc_nids; + spec->capsrc_nids = spec->private_capsrc_nids; + spec->num_adc_nids = 2; + spec->dual_adc_switch = 1; + snd_printdd("realtek: enabling dual ADC switchg (%02x:%02x)\n", + spec->adc_nids[0], spec->adc_nids[1]); + return 1; + } + return 0; +} + /* * BIOS auto configuration */ @@ -14149,8 +14377,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -14165,13 +14392,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - fillup_priv_adc_nids(codec, alc269_adc_candidates, - sizeof(alc269_adc_candidates)); + + if (!alc275_setup_dual_adc(codec)) + fillup_priv_adc_nids(codec, alc269_adc_candidates, + sizeof(alc269_adc_candidates)); /* set default input source */ - snd_hda_codec_write_cache(codec, spec->capsrc_nids[0], - 0, AC_VERB_SET_CONNECT_SEL, - spec->input_mux->items[0].index); + if (!spec->dual_adc_switch) + select_or_unmute_capsrc(codec, spec->capsrc_nids[0], + spec->input_mux->items[0].index); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -14195,6 +14424,7 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_multi_out(codec); alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -14450,10 +14680,12 @@ static int patch_alc269(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; + if (has_cdefine_beep(codec)) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } } if (board_config != ALC269_AUTO) @@ -14465,6 +14697,10 @@ static int patch_alc269(struct hda_codec *codec) */ spec->stream_analog_playback = &alc269_44k_pcm_analog_playback; spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; + } else if (spec->dual_adc_switch) { + spec->stream_analog_playback = &alc269_pcm_analog_playback; + /* switch ADC dynamically */ + spec->stream_analog_capture = &dualmic_pcm_analog_capture; } else { spec->stream_analog_playback = &alc269_pcm_analog_playback; spec->stream_analog_capture = &alc269_pcm_analog_capture; @@ -14486,7 +14722,7 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(codec); - if (spec->cdefine.enable_pcbeep) + if (has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); if (board_config == ALC269_AUTO) @@ -15350,8 +15586,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -15377,6 +15612,7 @@ static void alc861_auto_init(struct hda_codec *codec) alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -16481,8 +16717,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -16509,6 +16744,7 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc861vd_auto_init_hp_out(codec); alc861vd_auto_init_analog_input(codec); alc861vd_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -18492,7 +18728,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t dac) { int i, num; - hda_nid_t srcs[4]; + hda_nid_t srcs[HDA_MAX_CONNECTIONS]; alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ @@ -18596,8 +18832,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -18607,7 +18842,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc662_init_verbs); if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) add_verb(spec, alc663_init_verbs); if (codec->vendor_id == 0x10ec0272) @@ -18634,6 +18869,7 @@ static void alc662_auto_init(struct hda_codec *codec) alc662_auto_init_hp_out(codec); alc662_auto_init_analog_input(codec); alc662_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -18683,10 +18919,12 @@ static int patch_alc662(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; + if (has_cdefine_beep(codec)) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } } if (board_config != ALC662_AUTO) @@ -18708,7 +18946,7 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(codec); - if (spec->cdefine.enable_pcbeep) { + if (has_cdefine_beep(codec)) { switch (codec->vendor_id) { case 0x10ec0662: set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); @@ -18751,6 +18989,333 @@ static int patch_alc888(struct hda_codec *codec) } /* + * ALC680 support + */ +#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID +#define alc680_modes alc260_modes + +static hda_nid_t alc680_dac_nids[3] = { + /* Lout1, Lout2, hp */ + 0x02, 0x03, 0x04 +}; + +static hda_nid_t alc680_adc_nids[3] = { + /* ADC0-2 */ + /* DMIC, MIC, Line-in*/ + 0x07, 0x08, 0x09 +}; + +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("Mic Boost", 0x18, 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), + { } /* end */ +}; + +/* + * 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}, + + {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}, + + {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}, + { } +}; + +/* 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) +{ + hda_nid_t dac; + int err; + + switch (nid) { + case 0x14: + dac = 0x02; + break; + case 0x15: + dac = 0x03; + break; + case 0x16: + dac = 0x04; + break; + default: + return 0; + } + if (spec->multiout.dac_nids[0] != dac && + spec->multiout.dac_nids[1] != dac) { + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, + HDA_COMPOSE_AMP_VAL(dac, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, + HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); + + if (err < 0) + return err; + spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int alc680_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.dac_nids = spec->private_dac_nids; + + nid = cfg->line_out_pins[0]; + if (nid) { + const char *name; + if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + name = "Speaker"; + else + name = "Front"; + err = alc680_new_analog_output(spec, nid, name, 0); + if (err < 0) + return err; + } + + nid = cfg->speaker_pins[0]; + if (nid) { + err = alc680_new_analog_output(spec, nid, "Speaker", 0); + if (err < 0) + return err; + } + nid = cfg->hp_pins[0]; + if (nid) { + err = alc680_new_analog_output(spec, nid, "Headphone", 0); + if (err < 0) + return err; + } + + return 0; +} + +static void alc680_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type) +{ + alc_set_pin_output(codec, nid, pin_type); +} + +static void alc680_auto_init_multi_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid = spec->autocfg.line_out_pins[0]; + if (nid) { + int pin_type = get_pin_type(spec->autocfg.line_out_type); + alc680_auto_set_output_and_unmute(codec, nid, pin_type); + } +} + +static void alc680_auto_init_hp_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + + pin = spec->autocfg.hp_pins[0]; + if (pin) + alc680_auto_set_output_and_unmute(codec, pin, PIN_HP); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc680_auto_set_output_and_unmute(codec, pin, PIN_OUT); +} + +/* pcm configuration: identical with ALC880 */ +#define alc680_pcm_analog_playback alc880_pcm_analog_playback +#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 }, + }, +}; + +/* + * BIOS auto configuration + */ +static int alc680_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc680_ignore[] = { 0 }; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + 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; + spec->no_analog = 1; + goto dig_only; + } + return 0; /* can't find valid BIOS pin config */ + } + err = alc680_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = 2; + + dig_only: + /* digital only support output */ + alc_auto_parse_digital(codec); + if (spec->kctls.list) + 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) + return err; + + return 1; +} + +#define alc680_auto_init_analog_input alc882_auto_init_analog_input + +/* init callback for auto-configuration model -- overriding the default init */ +static void alc680_auto_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + alc680_auto_init_multi_out(codec); + alc680_auto_init_hp_out(codec); + alc680_auto_init_analog_input(codec); + alc_auto_init_digital(codec); + if (spec->unsol_event) + alc_inithook(codec); +} + +/* + * configuration and preset + */ +static const char *alc680_models[ALC680_MODEL_LAST] = { + [ALC680_BASE] = "base", + [ALC680_AUTO] = "auto", +}; + +static struct snd_pci_quirk alc680_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE), + {} +}; + +static struct alc_config_preset alc680_presets[] = { + [ALC680_BASE] = { + .mixers = { alc680_base_mixer }, + .cap_mixer = alc680_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, + }, +}; + +static int patch_alc680(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ALC680_MODEL_LAST, + alc680_models, + alc680_cfg_tbl); + + if (board_config < 0 || board_config >= ALC680_MODEL_LAST) { + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + board_config = ALC680_AUTO; + } + + if (board_config == ALC680_AUTO) { + /* automatic parse from the BIOS config */ + err = alc680_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC680_BASE; + } + } + + if (board_config != ALC680_AUTO) + 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_digital_playback = &alc680_pcm_digital_playback; + + if (!spec->adc_nids) { + spec->adc_nids = alc680_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc680_adc_nids); + } + + if (!spec->cap_mixer) + set_capture_mixer(codec); + + spec->vmaster_nid = 0x02; + + codec->patch_ops = alc_patch_ops; + if (board_config == ALC680_AUTO) + spec->init_hook = alc680_auto_init; + + return 0; +} + +/* * patch entries */ static struct hda_codec_preset snd_hda_preset_realtek[] = { @@ -18774,6 +19339,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f1e7babd6920..b8d730c47df1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -202,6 +202,7 @@ struct sigmatel_spec { unsigned int spdif_mute: 1; unsigned int check_volume_offset:1; unsigned int auto_mic:1; + unsigned int linear_tone_beep:1; /* gpio lines */ unsigned int eapd_mask; @@ -3802,7 +3803,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return err; if (codec->beep) { /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = 1; + codec->beep->linear_tone = spec->linear_tone_beep; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (!(caps & AC_AMPCAP_MUTE)) { @@ -5005,6 +5006,7 @@ static int patch_stac9200(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, @@ -5068,6 +5070,7 @@ static int patch_stac925x(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; @@ -5153,6 +5156,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 0; codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); spec->pin_nids = stac92hd73xx_pin_nids; @@ -5300,6 +5304,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; spec->mux_nids = stac92hd83xxx_mux_nids; @@ -5522,6 +5527,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 0; codec->patch_ops = stac92xx_patch_ops; spec->num_pins = STAC92HD71BXX_NUM_PINS; switch (codec->vendor_id) { @@ -5779,6 +5785,7 @@ static int patch_stac922x(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, @@ -5883,6 +5890,7 @@ static int patch_stac927x(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; @@ -6018,6 +6026,7 @@ static int patch_stac9205(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS, @@ -6174,6 +6183,7 @@ static int patch_stac9872(struct hda_codec *codec) return -ENOMEM; codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); spec->pin_nids = stac9872_pin_nids; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 73453814e098..ae3acb2b42d1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -552,24 +552,30 @@ static void via_auto_init_hp_out(struct hda_codec *codec) } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void via_auto_init_analog_input(struct hda_codec *codec) { struct via_spec *spec = codec->spec; + unsigned int ctl; int i; for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; + if (!nid) + continue; + if (spec->smart51_enabled && is_smart51_pins(spec, nid)) + ctl = PIN_OUT; + else if (i <= AUTO_PIN_FRONT_MIC) + ctl = PIN_VREF50; + else + ctl = PIN_IN; snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF50 : PIN_IN)); - + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); } } -static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); - static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -658,6 +664,8 @@ static void set_jack_power_state(struct hda_codec *codec) /* PW0 (19h), SW1 (18h), AOW1 (11h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + parm = AC_PWRST_D0; snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, @@ -667,6 +675,8 @@ static void set_jack_power_state(struct hda_codec *codec) if (is_8ch) { parm = AC_PWRST_D3; set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + parm = AC_PWRST_D0; snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x24, 0, @@ -3915,6 +3925,13 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, } } + /* for Smart 5.1, line/mic inputs double as output pins */ + if (cfg->line_outs == 1) { + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; + } + return 0; } @@ -3932,7 +3949,8 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + /* for Smart 5.1, there are always at least six channels */ + if (!nid && i > AUTO_SEQ_CENLFE) continue; nid_vol = nid_vols[i]; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 22208b373fb9..e1ec6d91ea38 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -73,7 +73,7 @@ struct psc_dma { }; /* Utility for retrieving psc_dma_stream structure from a substream */ -inline struct psc_dma_stream * +static inline struct psc_dma_stream * to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) { if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1941a357e8c4..d256f5f313b5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -328,38 +328,6 @@ static struct snd_soc_device spitz_snd_devdata = { .codec_dev = &soc_codec_dev_wm8750, }; -/* - * FIXME: This is a temporary bodge to avoid cross-tree merge issues. - * New drivers should register the wm8750 I2C device in the machine - * setup code (under arch/arm for ARM systems). - */ -static int wm8750_i2c_register(void) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = 0x1b; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(0); - if (!adapter) { - printk(KERN_ERR "can't get i2c adapter 0\n"); - return -ENODEV; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - printk(KERN_ERR "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - return -ENODEV; - } - - return 0; -} - static struct platform_device *spitz_snd_device; static int __init spitz_init(void) @@ -369,10 +337,6 @@ static int __init spitz_init(void) if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) return -ENODEV; - ret = wm8750_i2c_setup(); - if (ret != 0) - return ret; - spitz_snd_device = platform_device_alloc("soc-audio", -1); if (!spitz_snd_device) return -ENOMEM; diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4c7b051f9d17..1bc56b2b94e2 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -69,7 +69,6 @@ struct snd_at73c213 { int irq; int period; unsigned long bitrate; - struct clk *bitclk; struct ssc_device *ssc; struct spi_device *spi; u8 spi_wbuffer[2]; diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e7ac7f493a8f..1e362bf8834f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,7 +11,8 @@ snd-usb-audio-objs := card.o \ endpoint.o \ urb.o \ pcm.o \ - helper.o + helper.o \ + clock.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/card.c b/sound/usb/card.c index da1346bd4856..7a8ac1d81be7 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -236,7 +236,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } case UAC_VERSION_2: { - struct uac_clock_source_descriptor *cs; struct usb_interface_assoc_descriptor *assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc; @@ -245,21 +244,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) return -EINVAL; } - /* FIXME: for now, we expect there is at least one clock source - * descriptor and we always take the first one. - * We should properly support devices with multiple clock sources, - * clock selectors and sample rate conversion units. */ - - cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, UAC2_CLOCK_SOURCE); - - if (!cs) { - snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); - return -EINVAL; - } - - chip->clock_id = cs->bClockID; - for (i = 0; i < assoc->bInterfaceCount; i++) { int intf = assoc->bFirstInterface + i; @@ -481,6 +465,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } + chip->ctrl_intf = alts; + if (err > 0) { /* create normal USB audio interfaces */ if (snd_usb_create_streams(chip, ifnum) < 0 || diff --git a/sound/usb/card.h b/sound/usb/card.h index ed92420c1095..1febf2f23754 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -25,6 +25,7 @@ struct audioformat { unsigned int rate_min, rate_max; /* min/max rates */ unsigned int nr_rates; /* number of rate table entries */ unsigned int *rate_table; /* rate table */ + unsigned char clock; /* associated clock */ }; struct snd_usb_substream; diff --git a/sound/usb/clock.c b/sound/usb/clock.c new file mode 100644 index 000000000000..b5855114667e --- /dev/null +++ b/sound/usb/clock.c @@ -0,0 +1,315 @@ +/* + * Clock domain and sample rate management functions + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/bitops.h> +#include <linux/init.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/string.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> + +#include "usbaudio.h" +#include "card.h" +#include "midi.h" +#include "mixer.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "helper.h" +#include "debug.h" +#include "pcm.h" +#include "urb.h" +#include "format.h" + +static struct uac_clock_source_descriptor * + snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_source_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_SOURCE))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static struct uac_clock_selector_descriptor * + snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_selector_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_SELECTOR))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static struct uac_clock_multiplier_descriptor * + snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_multiplier_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_MULTIPLIER))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) +{ + unsigned char buf; + int ret; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), + UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + UAC2_CX_CLOCK_SELECTOR << 8, + snd_usb_ctrl_intf(chip) | (selector_id << 8), + &buf, sizeof(buf), 1000); + + if (ret < 0) + return ret; + + return buf; +} + +static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) +{ + int err; + unsigned char data; + struct usb_device *dev = chip->dev; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, + snd_usb_ctrl_intf(chip) | (source_id << 8), + &data, sizeof(data), 1000); + + if (err < 0) { + snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return err; + } + + return !!data; +} + +/* Try to find the clock source ID of a given clock entity */ + +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id, unsigned long *visited) +{ + struct uac_clock_source_descriptor *source; + struct uac_clock_selector_descriptor *selector; + struct uac_clock_multiplier_descriptor *multiplier; + + entity_id &= 0xff; + + if (test_and_set_bit(entity_id, visited)) { + snd_printk(KERN_WARNING + "%s(): recursive clock topology detected, id %d.\n", + __func__, entity_id); + return -EINVAL; + } + + /* first, see if the ID we're looking for is a clock source already */ + source = snd_usb_find_clock_source(host_iface, entity_id); + if (source) + return source->bClockID; + + selector = snd_usb_find_clock_selector(host_iface, entity_id); + if (selector) { + int ret; + + /* the entity ID we are looking for is a selector. + * find out what it currently selects */ + ret = uac_clock_selector_get_val(chip, selector->bClockID); + if (ret < 0) + return ret; + + if (ret > selector->bNrInPins || ret < 1) { + printk(KERN_ERR + "%s(): selector reported illegal value, id %d, ret %d\n", + __func__, selector->bClockID, ret); + + return -EINVAL; + } + + return __uac_clock_find_source(chip, host_iface, + selector->baCSourceID[ret-1], + visited); + } + + /* FIXME: multipliers only act as pass-thru element for now */ + multiplier = snd_usb_find_clock_multiplier(host_iface, entity_id); + if (multiplier) + return __uac_clock_find_source(chip, host_iface, + multiplier->bCSourceID, visited); + + return -EINVAL; +} + +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id) +{ + DECLARE_BITMAP(visited, 256); + memset(visited, 0, sizeof(visited)); + return __uac_clock_find_source(chip, host_iface, entity_id, visited); +} + +static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned int ep; + unsigned char data[3]; + int err, crate; + + ep = get_endpoint(alts, 0)->bEndpointAddress; + + /* if endpoint doesn't have sampling rate control, bail out */ + if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { + snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", + dev->devnum, iface, fmt->altsetting); + return 0; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", + dev->devnum, iface, fmt->altsetting, rate, ep); + return err; + } + + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", + dev->devnum, iface, fmt->altsetting, ep); + return 0; /* some devices don't support reading */ + } + + crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (crate != rate) { + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + // runtime->rate = crate; + } + + return 0; +} + +static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned char data[4]; + int err, crate; + int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fmt->clock); + + if (clock < 0) + return clock; + + if (!uac_clock_source_is_valid(chip, clock)) { + snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n", + dev->devnum, iface, fmt->altsetting, clock); + return -ENXIO; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + data[3] = rate >> 24; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", + dev->devnum, iface, fmt->altsetting, rate); + return err; + } + + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (crate != rate) + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + + return 0; +} + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + + switch (altsd->bInterfaceProtocol) { + case UAC_VERSION_1: + return set_sample_rate_v1(chip, iface, alts, fmt, rate); + + case UAC_VERSION_2: + return set_sample_rate_v2(chip, iface, alts, fmt, rate); + } + + return -EINVAL; +} + diff --git a/sound/usb/clock.h b/sound/usb/clock.h new file mode 100644 index 000000000000..beb253684e2d --- /dev/null +++ b/sound/usb/clock.h @@ -0,0 +1,12 @@ +#ifndef __USBAUDIO_CLOCK_H +#define __USBAUDIO_CLOCK_H + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate); + +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id); + +#endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 28ee1ce3971a..6f6596cf2b19 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -190,6 +190,38 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, return attributes; } +static struct uac2_input_terminal_descriptor * + snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_input_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_INPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +static struct uac2_output_terminal_descriptor * + snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_output_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_OUTPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -199,7 +231,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) int i, altno, err, stream; int format = 0, num_channels = 0; struct audioformat *fp = NULL; - int num, protocol; + int num, protocol, clock = 0; struct uac_format_type_i_continuous_descriptor *fmt; dev = chip->dev; @@ -263,6 +295,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } case UAC_VERSION_2: { + struct uac2_input_terminal_descriptor *input_term; + struct uac2_output_terminal_descriptor *output_term; struct uac_as_header_descriptor_v2 *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -281,7 +315,25 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num_channels = as->bNrChannels; format = le32_to_cpu(as->bmFormats); - break; + /* lookup the terminal associated to this interface + * to extract the clock */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + break; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + break; + } + + snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", + dev->devnum, iface_no, altno, as->bTerminalLink); + continue; } default: @@ -338,6 +390,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); + fp->clock = clock; /* some quirks for attributes here */ @@ -374,6 +427,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { kfree(fp->rate_table); kfree(fp); + fp = NULL; continue; } diff --git a/sound/usb/format.c b/sound/usb/format.c index fe29d61de19b..30364aba79cc 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -29,6 +29,7 @@ #include "quirks.h" #include "helper.h" #include "debug.h" +#include "clock.h" /* * parse the audio format type I descriptor @@ -205,6 +206,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof } /* + * Helper function to walk the array of sample rate triplets reported by + * the device. The problem is that we need to parse whole array first to + * get to know how many sample rates we have to expect. + * Then fp->rate_table can be allocated and filled. + */ +static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, + const unsigned char *data) +{ + int i, nr_rates = 0; + + fp->rates = fp->rate_min = fp->rate_max = 0; + + for (i = 0; i < nr_triplets; i++) { + int min = combine_quad(&data[2 + 12 * i]); + int max = combine_quad(&data[6 + 12 * i]); + int res = combine_quad(&data[10 + 12 * i]); + int rate; + + if ((max < 0) || (min < 0) || (res < 0) || (max < min)) + continue; + + /* + * for ranges with res == 1, we announce a continuous sample + * rate range, and this function should return 0 for no further + * parsing. + */ + if (res == 1) { + fp->rate_min = min; + fp->rate_max = max; + fp->rates = SNDRV_PCM_RATE_CONTINUOUS; + return 0; + } + + for (rate = min; rate <= max; rate += res) { + if (fp->rate_table) + fp->rate_table[nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + + nr_rates++; + + /* avoid endless loop */ + if (res == 0) + break; + } + } + + return nr_rates; +} + +/* * parse the format descriptor and stores the possible sample rates * on the audioformat table (audio class v2). */ @@ -214,21 +269,30 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, { struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; - int i, nr_rates, data_size, ret = 0; + int nr_triplets, data_size, ret = 0; + int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock); + + if (clock < 0) { + snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n", + __func__, clock); + goto err; + } /* get the number of sample rates first by only fetching 2 bytes */ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), tmp, sizeof(tmp), 1000); if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); goto err; } - nr_rates = (tmp[1] << 8) | tmp[0]; - data_size = 2 + 12 * nr_rates; + nr_triplets = (tmp[1] << 8) | tmp[0]; + data_size = 2 + 12 * nr_triplets; data = kzalloc(data_size, GFP_KERNEL); if (!data) { ret = -ENOMEM; @@ -237,36 +301,40 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, /* now get the full information */ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, data_size, 1000); + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, data_size, 1000); if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", + __func__, clock); ret = -EINVAL; goto err_free; } - fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + /* Call the triplet parser, and make sure fp->rate_table is NULL. + * We just use the return value to know how many sample rates we + * will have to deal with. */ + kfree(fp->rate_table); + fp->rate_table = NULL; + fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data); + + if (fp->nr_rates == 0) { + /* SNDRV_PCM_RATE_CONTINUOUS */ + ret = 0; + goto err_free; + } + + fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); if (!fp->rate_table) { ret = -ENOMEM; goto err_free; } - fp->nr_rates = 0; - fp->rate_min = fp->rate_max = 0; - - for (i = 0; i < nr_rates; i++) { - int rate = combine_quad(&data[2 + 12 * i]); - - fp->rate_table[fp->nr_rates] = rate; - if (!fp->rate_min || rate < fp->rate_min) - fp->rate_min = rate; - if (!fp->rate_max || rate > fp->rate_max) - fp->rate_max = rate; - fp->rates |= snd_pcm_rate_to_rate_bit(rate); - fp->nr_rates++; - } + /* Call the triplet parser again, but this time, fp->rate_table is + * allocated, so the rates will be stored */ + parse_uac2_sample_rate_range(fp, nr_triplets, data); err_free: kfree(data); diff --git a/sound/usb/helper.h b/sound/usb/helper.h index a6b0e51b3a9a..09bd943c43bf 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -28,5 +28,9 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, #define snd_usb_get_speed(dev) ((dev)->speed) #endif +static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip) +{ + return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber; +} #endif /* __USBAUDIO_HELPER_H */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 03ce971e0027..736d134cc03c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -78,39 +78,6 @@ enum { USB_MIXER_U16, }; -enum { - USB_PROC_UPDOWN = 1, - USB_PROC_UPDOWN_SWITCH = 1, - USB_PROC_UPDOWN_MODE_SEL = 2, - - USB_PROC_PROLOGIC = 2, - USB_PROC_PROLOGIC_SWITCH = 1, - USB_PROC_PROLOGIC_MODE_SEL = 2, - - USB_PROC_3DENH = 3, - USB_PROC_3DENH_SWITCH = 1, - USB_PROC_3DENH_SPACE = 2, - - USB_PROC_REVERB = 4, - USB_PROC_REVERB_SWITCH = 1, - USB_PROC_REVERB_LEVEL = 2, - USB_PROC_REVERB_TIME = 3, - USB_PROC_REVERB_DELAY = 4, - - USB_PROC_CHORUS = 5, - USB_PROC_CHORUS_SWITCH = 1, - USB_PROC_CHORUS_LEVEL = 2, - USB_PROC_CHORUS_RATE = 3, - USB_PROC_CHORUS_DEPTH = 4, - - USB_PROC_DCR = 6, - USB_PROC_DCR_SWITCH = 1, - USB_PROC_DCR_RATIO = 2, - USB_PROC_DCR_MAX_AMP = 3, - USB_PROC_DCR_THRESHOLD = 4, - USB_PROC_DCR_ATTACK = 5, - USB_PROC_DCR_RELEASE = 6, -}; /*E-mu 0202(0404) eXtension Unit(XU) control*/ enum { @@ -198,22 +165,24 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, /* * find an audio control unit with the given unit id - * this doesn't return any clock related units, so they need to be handled elsewhere */ static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit) { - unsigned char *p; + /* we just parse the header */ + struct uac_feature_unit_descriptor *hdr = NULL; - p = NULL; - while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, - USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC2_EXTENSION_UNIT_V2 && p[3] == unit) - return p; + while ((hdr = snd_usb_find_desc(state->buffer, state->buflen, hdr, + USB_DT_CS_INTERFACE)) != NULL) { + if (hdr->bLength >= 4 && + hdr->bDescriptorSubtype >= UAC_INPUT_TERMINAL && + hdr->bDescriptorSubtype <= UAC2_SAMPLE_RATE_CONVERTER && + hdr->bUnitID == unit) + return hdr; } + return NULL; } - /* * copy a string with the given id */ @@ -328,27 +297,36 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { - unsigned char buf[14]; /* enough space for one range of 4 bytes */ + unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */ unsigned char *val; - int ret; + int ret, size; __u8 bRequest; - bRequest = (request == UAC_GET_CUR) ? - UAC2_CS_CUR : UAC2_CS_RANGE; + if (request == UAC_GET_CUR) { + bRequest = UAC2_CS_CUR; + size = sizeof(__u16); + } else { + bRequest = UAC2_CS_RANGE; + size = sizeof(buf); + } + + memset(buf, 0, sizeof(buf)); ret = snd_usb_ctl_msg(cval->mixer->chip->dev, usb_rcvctrlpipe(cval->mixer->chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, cval->mixer->ctrlif | (cval->id << 8), - buf, sizeof(buf), 1000); + buf, size, 1000); if (ret < 0) { - snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); + snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", + request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); return ret; } + /* FIXME: how should we handle multiple triplets here? */ + switch (request) { case UAC_GET_CUR: val = buf; @@ -462,6 +440,16 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; + unsigned int read_only = (channel == 0) ? + cval->master_readonly : + cval->ch_readonly & (1 << (channel - 1)); + + if (read_only) { + snd_printdd(KERN_INFO "%s(): channel %d of control %d is read_only\n", + __func__, channel, cval->control); + return 0; + } + err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) @@ -631,6 +619,7 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm */ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { + int err; void *p1; memset(term, 0, sizeof(*term)); @@ -651,6 +640,11 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->channels = d->bNrChannels; term->chconfig = le32_to_cpu(d->bmChannelConfig); term->name = d->iTerminal; + + /* call recursively to get the clock selectors */ + err = check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; } return 0; case UAC_FEATURE_UNIT: { @@ -667,7 +661,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->name = uac_mixer_unit_iMixer(d); return 0; } - case UAC_SELECTOR_UNIT: { + case UAC_SELECTOR_UNIT: + case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ if (check_input_term(state, d->baSourceID[0], term) < 0) @@ -690,6 +685,13 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->name = uac_processing_unit_iProcessing(d, state->mixer->protocol); return 0; } + case UAC2_CLOCK_SOURCE: { + struct uac_clock_source_descriptor *d = p1; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = d->iClockSource; + return 0; + } default: return -ENODEV; } @@ -709,16 +711,20 @@ struct usb_feature_control_info { }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, + { "Mute", USB_MIXER_INV_BOOLEAN }, + { "Volume", USB_MIXER_S16 }, { "Tone Control - Bass", USB_MIXER_S8 }, { "Tone Control - Mid", USB_MIXER_S8 }, { "Tone Control - Treble", USB_MIXER_S8 }, { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Auto Gain Control", USB_MIXER_BOOLEAN }, + { "Delay Control", USB_MIXER_U16 }, + { "Bass Boost", USB_MIXER_BOOLEAN }, + { "Loudness", USB_MIXER_BOOLEAN }, + /* UAC2 specific */ + { "Input Gain Control", USB_MIXER_U16 }, + { "Input Gain Pad Control", USB_MIXER_BOOLEAN }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; @@ -958,7 +964,7 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) static void build_feature_ctl(struct mixer_build *state, void *raw_desc, unsigned int ctl_mask, int control, struct usb_audio_term *iterm, int unitid, - int read_only) + int readonly_mask) { struct uac_feature_unit_descriptor *desc = raw_desc; unsigned int len = 0; @@ -970,7 +976,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, control++; /* change from zero-based to 1-based value */ - if (control == UAC_GRAPHIC_EQUALIZER_CONTROL) { + if (control == UAC_FU_GRAPHIC_EQUALIZER) { /* FIXME: not supported yet */ return; } @@ -989,20 +995,25 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->control = control; cval->cmask = ctl_mask; cval->val_type = audio_feature_info[control-1].type; - if (ctl_mask == 0) + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ - else { + cval->master_readonly = readonly_mask; + } else { int i, c = 0; for (i = 0; i < 16; i++) if (ctl_mask & (1 << i)) c++; cval->channels = c; + cval->ch_readonly = readonly_mask; } /* get min/max values */ get_min_max(cval, 0); - if (read_only) + /* if all channels in the mask are marked read-only, make the control + * read-only. set_cur_mix_value() will check the mask again and won't + * issue write commands to read-only channels. */ + if (cval->channels == readonly_mask) kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); else kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); @@ -1021,8 +1032,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kctl->id.name, sizeof(kctl->id.name)); switch (control) { - case UAC_MUTE_CONTROL: - case UAC_VOLUME_CONTROL: + case UAC_FU_MUTE: + case UAC_FU_VOLUME: /* determine the control name. the rule is: * - if a name id is given in descriptor, use it. * - if the connected input can be determined, then use the name @@ -1049,9 +1060,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = append_ctl_name(kctl, " Playback"); } } - append_ctl_name(kctl, control == UAC_MUTE_CONTROL ? + append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); - if (control == UAC_VOLUME_CONTROL) { + if (control == UAC_FU_VOLUME) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | @@ -1096,6 +1107,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } break; + case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0991): + /* Most audio usb devices lie about volume resolution. + * Most Logitech webcams have res = 384. + * Proboly there is some logitech magic behind this number --fishor + */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set resolution quirk: cval->res = 384\n"); + cval->res = 384; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", @@ -1150,7 +1174,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void snd_printk(KERN_INFO "usbmixer: master volume quirk for PCM2702 chip\n"); /* disable non-functional volume control */ - master_bits &= ~UAC_FU_VOLUME; + master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); break; } if (channels > 0) @@ -1188,19 +1212,22 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void for (j = 0; j < channels; j++) { unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); - if (mask & (1 << (i * 2))) { + if (uac2_control_is_readable(mask, i)) { ch_bits |= (1 << j); - if (~mask & (1 << ((i * 2) + 1))) + if (!uac2_control_is_writeable(mask, i)) ch_read_only |= (1 << j); } } - /* FIXME: the whole unit is read-only if any of the channels is marked read-only */ + /* NOTE: build_feature_ctl() will mark the control read-only if all channels + * are marked read-only in the descriptors. Otherwise, the control will be + * reported as writeable, but the driver will not actually issue a write + * command for read-only channels */ if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only); - if (master_bits & (1 << i * 2)) + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); + if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, - ~master_bits & (1 << ((i * 2) + 1))); + !uac2_control_is_writeable(master_bits, i)); } } @@ -1392,51 +1419,51 @@ struct procunit_info { }; static struct procunit_value_info updown_proc_info[] = { - { USB_PROC_UPDOWN_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 }, + { UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; static struct procunit_value_info prologic_proc_info[] = { - { USB_PROC_PROLOGIC_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 }, + { UAC_DP_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_DP_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; static struct procunit_value_info threed_enh_proc_info[] = { - { USB_PROC_3DENH_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_3DENH_SPACE, "Spaciousness", USB_MIXER_U8 }, + { UAC_3D_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_3D_SPACE, "Spaciousness", USB_MIXER_U8 }, { 0 } }; static struct procunit_value_info reverb_proc_info[] = { - { USB_PROC_REVERB_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_REVERB_LEVEL, "Level", USB_MIXER_U8 }, - { USB_PROC_REVERB_TIME, "Time", USB_MIXER_U16 }, - { USB_PROC_REVERB_DELAY, "Delay", USB_MIXER_U8 }, + { UAC_REVERB_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_REVERB_LEVEL, "Level", USB_MIXER_U8 }, + { UAC_REVERB_TIME, "Time", USB_MIXER_U16 }, + { UAC_REVERB_FEEDBACK, "Feedback", USB_MIXER_U8 }, { 0 } }; static struct procunit_value_info chorus_proc_info[] = { - { USB_PROC_CHORUS_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_CHORUS_LEVEL, "Level", USB_MIXER_U8 }, - { USB_PROC_CHORUS_RATE, "Rate", USB_MIXER_U16 }, - { USB_PROC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 }, + { UAC_CHORUS_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_CHORUS_LEVEL, "Level", USB_MIXER_U8 }, + { UAC_CHORUS_RATE, "Rate", USB_MIXER_U16 }, + { UAC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 }, { 0 } }; static struct procunit_value_info dcr_proc_info[] = { - { USB_PROC_DCR_SWITCH, "Switch", USB_MIXER_BOOLEAN }, - { USB_PROC_DCR_RATIO, "Ratio", USB_MIXER_U16 }, - { USB_PROC_DCR_MAX_AMP, "Max Amp", USB_MIXER_S16 }, - { USB_PROC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 }, - { USB_PROC_DCR_ATTACK, "Attack Time", USB_MIXER_U16 }, - { USB_PROC_DCR_RELEASE, "Release Time", USB_MIXER_U16 }, + { UAC_DCR_ENABLE, "Switch", USB_MIXER_BOOLEAN }, + { UAC_DCR_RATE, "Ratio", USB_MIXER_U16 }, + { UAC_DCR_MAXAMPL, "Max Amp", USB_MIXER_S16 }, + { UAC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 }, + { UAC_DCR_ATTACK_TIME, "Attack Time", USB_MIXER_U16 }, + { UAC_DCR_RELEASE_TIME, "Release Time", USB_MIXER_U16 }, { 0 } }; static struct procunit_info procunits[] = { - { USB_PROC_UPDOWN, "Up Down", updown_proc_info }, - { USB_PROC_PROLOGIC, "Dolby Prologic", prologic_proc_info }, - { USB_PROC_3DENH, "3D Stereo Extender", threed_enh_proc_info }, - { USB_PROC_REVERB, "Reverb", reverb_proc_info }, - { USB_PROC_CHORUS, "Chorus", chorus_proc_info }, - { USB_PROC_DCR, "DCR", dcr_proc_info }, + { UAC_PROCESS_UP_DOWNMIX, "Up Down", updown_proc_info }, + { UAC_PROCESS_DOLBY_PROLOGIC, "Dolby Prologic", prologic_proc_info }, + { UAC_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", threed_enh_proc_info }, + { UAC_PROCESS_REVERB, "Reverb", reverb_proc_info }, + { UAC_PROCESS_CHORUS, "Chorus", chorus_proc_info }, + { UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info }, { 0 }, }; /* @@ -1524,7 +1551,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->channels = 1; /* get min/max values */ - if (type == USB_PROC_UPDOWN && cval->control == USB_PROC_UPDOWN_MODE_SEL) { + if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) { __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol); /* FIXME: hard-coded */ cval->min = 1; @@ -1619,7 +1646,7 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, err; - err = get_cur_ctl_value(cval, 0, &val); + err = get_cur_ctl_value(cval, cval->control << 8, &val); if (err < 0) { if (cval->mixer->ignore_ctl_error) { ucontrol->value.enumerated.item[0] = 0; @@ -1638,7 +1665,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, oval, err; - err = get_cur_ctl_value(cval, 0, &oval); + err = get_cur_ctl_value(cval, cval->control << 8, &oval); if (err < 0) { if (cval->mixer->ignore_ctl_error) return 0; @@ -1647,7 +1674,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ val = ucontrol->value.enumerated.item[0]; val = get_abs_value(cval, val); if (val != oval) { - set_cur_ctl_value(cval, 0, val); + set_cur_ctl_value(cval, cval->control << 8, val); return 1; } return 0; @@ -1729,6 +1756,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void cval->res = 1; cval->initialized = 1; + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) + cval->control = UAC2_CX_CLOCK_SELECTOR; + else + cval->control = 0; + namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); if (! namelist) { snd_printk(KERN_ERR "cannot malloc\n"); @@ -1778,7 +1810,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void if (! len) strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); - if ((state->oterm.type & 0xff00) == 0x0100) + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) + append_ctl_name(kctl, " Clock Source"); + else if ((state->oterm.type & 0xff00) == 0x0100) append_ctl_name(kctl, " Capture Source"); else append_ctl_name(kctl, " Playback Source"); @@ -1812,10 +1846,12 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: + case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: + case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); @@ -1912,6 +1948,11 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) return err; + + /* for UAC2, use the same approach to also add the clock selectors */ + err = parse_audio_unit(&state, desc->bCSourceID); + if (err < 0) + return err; } } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 130123854a6c..a7cf1007fbb0 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -34,6 +34,8 @@ struct usb_mixer_elem_info { unsigned int id; unsigned int control; /* CS or ICN (high byte) */ unsigned int cmask; /* channel mask bitmap: 0 = master */ + unsigned int ch_readonly; + unsigned int master_readonly; int channels; int val_type; int min, max, res; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index d93fc89beba8..f1324c423835 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -85,8 +85,8 @@ static struct usbmix_name_map extigy_map[] = { /* 16: MU (w/o controls) */ { 17, NULL, 1 }, /* DISABLED: PU-switch (any effect?) */ { 17, "Channel Routing", 2 }, /* PU: mode select */ - { 18, "Tone Control - Bass", UAC_BASS_CONTROL }, /* FU */ - { 18, "Tone Control - Treble", UAC_TREBLE_CONTROL }, /* FU */ + { 18, "Tone Control - Bass", UAC_FU_BASS }, /* FU */ + { 18, "Tone Control - Treble", UAC_FU_TREBLE }, /* FU */ { 18, "Master Playback" }, /* FU; others */ /* 19: OT speaker */ /* 20: OT headphone */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 056587de7be4..456829882f40 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -31,6 +31,7 @@ #include "urb.h" #include "helper.h" #include "pcm.h" +#include "clock.h" /* * return the current pcm pointer. just based on the hwptr_done value. @@ -181,103 +182,6 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, return -EINVAL; } -static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_device *dev = chip->dev; - unsigned int ep; - unsigned char data[3]; - int err, crate; - - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint doesn't have sampling rate control, bail out */ - if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { - snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", - dev->devnum, iface, fmt->altsetting); - return 0; - } - - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", - dev->devnum, iface, fmt->altsetting, rate, ep); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", - dev->devnum, iface, fmt->altsetting, ep); - return 0; /* some devices don't support reading */ - } - crate = data[0] | (data[1] << 8) | (data[2] << 16); - if (crate != rate) { - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - // runtime->rate = crate; - } - - return 0; -} - -static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_device *dev = chip->dev; - unsigned char data[4]; - int err, crate; - - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - data[3] = rate >> 24; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", - dev->devnum, iface, fmt->altsetting, rate); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", - dev->devnum, iface, fmt->altsetting); - return err; - } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - - return 0; -} - -int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - - switch (altsd->bInterfaceProtocol) { - case UAC_VERSION_1: - return set_sample_rate_v1(chip, iface, alts, fmt, rate); - - case UAC_VERSION_2: - return set_sample_rate_v2(chip, iface, alts, fmt, rate); - } - - return -EINVAL; -} - /* * find a matching format and set up the interface */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 06ebf24d3a4d..24d3319cc34d 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -40,9 +40,6 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; - /* for audio class v2 */ - int clock_id; - struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; @@ -53,6 +50,8 @@ struct snd_usb_audio { int setup; /* from the 'device_setup' module param */ int nrpacks; /* from the 'nrpacks' module param */ int async_unlink; /* from the 'async_unlink' module param */ + + struct usb_host_interface *ctrl_intf; /* the audio control interface */ }; /* |