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author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-06-21 13:13:08 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-06-21 13:13:08 -0700 |
commit | 413318444fd5351f9858b9deb4e8ecaf8898ee05 (patch) | |
tree | c5ab72670ca792c800ca6c75e534c96df2cb80c7 | |
parent | d06063cc221fdefcab86589e79ddfdb7c0e14b63 (diff) | |
parent | 47166281d2dc9daf7da9a5ad88491ae94366e852 (diff) | |
download | linux-413318444fd5351f9858b9deb4e8ecaf8898ee05.tar.gz linux-413318444fd5351f9858b9deb4e8ecaf8898ee05.tar.bz2 linux-413318444fd5351f9858b9deb4e8ecaf8898ee05.zip |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add model=6530g option
ALSA: hda - Acer Inspire 6530G model for Realtek ALC888
ALSA: snd_usb_caiaq: fix legacy input streaming
ASoC: Kill BUS_ID_SIZE
ALSA: HDA - Correct trivial typos in comments.
ALSA: HDA - Name-fixes in code (tagra/targa)
ALSA: HDA - Add pci-quirk for MSI MS-7350 motherboard.
ALSA: hda - Fix memory leak at codec creation
-rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 1 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 134 | ||||
-rw-r--r-- | sound/soc/txx9/txx9aclc.c | 4 | ||||
-rw-r--r-- | sound/usb/caiaq/audio.c | 5 | ||||
-rw-r--r-- | sound/usb/caiaq/device.c | 2 |
6 files changed, 110 insertions, 38 deletions
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index de8e10a94103..0d8d23581c44 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -139,6 +139,7 @@ ALC883/888 acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) acer-aspire Acer Aspire 9810 acer-aspire-4930g Acer Aspire 4930G + acer-aspire-6530g Acer Aspire 6530G acer-aspire-8930g Acer Aspire 8930G medion Medion Laptops medion-md2 Medion MD2 diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 562403a23488..462e2cedaa6a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -972,8 +972,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_SUBSYSTEM_ID, 0); } - if (bus->modelname) - codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); /* power-up all before initialization */ hda_set_power_state(codec, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d22b26068014..bf4b78a74a8f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -224,6 +224,7 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -970,7 +971,7 @@ static void alc_automute_pin(struct hda_codec *codec) } } -#if 0 /* it's broken in some acses -- temporarily disabled */ +#if 0 /* it's broken in some cases -- temporarily disabled */ static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1170,7 +1171,7 @@ static int alc_subsystem_id(struct hda_codec *codec, /* invalid SSID, check the special NID pin defcfg instead */ /* - * 31~30 : port conetcivity + * 31~30 : port connectivity * 29~21 : reserve * 20 : PCBEEP input * 19~16 : Check sum (15:1) @@ -1471,6 +1472,25 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Bias voltage on for external mic port */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Enable speaker output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, +/* Enable headphone output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1564,25 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 2, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -6347,7 +6386,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { }; /* - * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic + * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ /* @@ -7047,7 +7086,7 @@ static struct hda_verb alc882_auto_init_verbs[] = { #define alc882_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc882_pcm_analog_playback alc880_pcm_analog_playback #define alc882_pcm_analog_capture alc880_pcm_analog_capture #define alc882_pcm_digital_playback alc880_pcm_digital_playback @@ -8068,7 +8107,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_tagra_mixer[] = { +static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8088,7 +8127,7 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { +static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8153,6 +8192,19 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -8417,7 +8469,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_tagra_verbs[] = { +static struct hda_verb alc883_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -8626,8 +8678,8 @@ static void alc883_medion_md2_init_hook(struct hda_codec *codec) } /* toggle speaker-output according to the hp-jack state */ -#define alc883_tagra_init_hook alc882_targa_init_hook -#define alc883_tagra_unsol_event alc882_targa_unsol_event +#define alc883_targa_init_hook alc882_targa_init_hook +#define alc883_targa_unsol_event alc882_targa_unsol_event static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { @@ -8957,7 +9009,7 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) #define alc883_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc883_pcm_analog_playback alc880_pcm_analog_playback #define alc883_pcm_analog_capture alc880_pcm_analog_capture #define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture @@ -8978,6 +9030,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", @@ -9021,7 +9074,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9069,6 +9122,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), @@ -9165,8 +9219,8 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, }, [ALC883_TARGA_DIG] = { - .mixers = { alc883_tagra_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9174,12 +9228,12 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_TARGA_2ch_DIG] = { - .mixers = { alc883_tagra_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .mixers = { alc883_targa_2ch_mixer}, + .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, @@ -9188,13 +9242,13 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_TARGA_8ch_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, - alc883_tagra_verbs }, + alc883_targa_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), @@ -9206,8 +9260,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_4ST_8ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, @@ -9255,6 +9309,24 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_6530G] = { + .mixers = { alc888_acer_aspire_6530_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_6530g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_acer_aspire_6530_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_4930g_init_hook, + }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -9361,7 +9433,7 @@ static struct alc_config_preset alc883_presets[] = { .init_hook = alc888_lenovo_ms7195_front_automute, }, [ALC883_HAIER_W66] = { - .mixers = { alc883_tagra_2ch_mixer}, + .mixers = { alc883_targa_2ch_mixer}, .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, @@ -11131,7 +11203,7 @@ static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { #define alc262_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc262_pcm_analog_playback alc880_pcm_analog_playback #define alc262_pcm_analog_capture alc880_pcm_analog_capture #define alc262_pcm_digital_playback alc880_pcm_digital_playback @@ -12286,7 +12358,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2); } -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc268_pcm_analog_playback alc880_pcm_analog_playback #define alc268_pcm_analog_capture alc880_pcm_analog_capture #define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture @@ -13197,7 +13269,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, #define alc269_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc269_pcm_analog_playback alc880_pcm_analog_playback #define alc269_pcm_analog_capture alc880_pcm_analog_capture #define alc269_pcm_digital_playback alc880_pcm_digital_playback @@ -14059,7 +14131,7 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec, alc861_toshiba_automute(codec); } -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc861_pcm_analog_playback alc880_pcm_analog_playback #define alc861_pcm_analog_capture alc880_pcm_analog_capture #define alc861_pcm_digital_playback alc880_pcm_digital_playback @@ -14582,7 +14654,7 @@ static hda_nid_t alc861vd_dac_nids[4] = { /* dac_nids for ALC660vd are in a different order - according to * Realtek's driver. - * This should probably tesult in a different mixer for 6stack models + * This should probably result in a different mixer for 6stack models * of ALC660vd codecs, but for now there is only 3stack mixer * - and it is the same as in 861vd. * adc_nids in ALC660vd are (is) the same as in 861vd @@ -15027,7 +15099,7 @@ static void alc861vd_dallas_init_hook(struct hda_codec *codec) #define alc861vd_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc861vd_pcm_analog_playback alc880_pcm_analog_playback #define alc861vd_pcm_analog_capture alc880_pcm_analog_capture #define alc861vd_pcm_digital_playback alc880_pcm_digital_playback @@ -15206,7 +15278,7 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin; pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front and use dac 0 */ + if (pin) /* connect to front and use dac 0 */ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); pin = spec->autocfg.speaker_pins[0]; if (pin) @@ -16669,7 +16741,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = { #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc662_pcm_analog_playback alc880_pcm_analog_playback #define alc662_pcm_analog_capture alc880_pcm_analog_capture #define alc662_pcm_digital_playback alc880_pcm_digital_playback diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index fa336616152e..938a58a5a244 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -297,9 +297,9 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, static bool filter(struct dma_chan *chan, void *param) { struct txx9aclc_dmadata *dmadata = param; - char devname[BUS_ID_SIZE + 2]; + char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */ - sprintf(devname, "%s.%d", dmadata->dma_res->name, + snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name, (int)dmadata->dma_res->start); if (strcmp(dev_name(chan->device->dev), devname) == 0) { chan->private = &dmadata->dma_slave; diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b14451342166..8f9b60c5d74c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; } else { - dev->period_in_count[index] = BYTES_PER_SAMPLE; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; + int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; + dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; + dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; } if (dev->streaming) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 22406245a98b..0e5db719de24 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," |