From f75dcc87feab791847605044311a4a8e9335da91 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Jun 2009 08:22:32 +0200 Subject: ALSA: hda - Fix memory leak at codec creation The codec->modelname field is allocated twice in snd_hda_codec_new(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 562403a2348..462e2cedaa6 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, -- cgit v1.2.3-18-g5258 From df01b8af5627b7e28c087559cbce4f359ce07c49 Mon Sep 17 00:00:00 2001 From: Sasha Alexandr Date: Tue, 16 Jun 2009 14:46:17 -0400 Subject: ALSA: HDA - Add pci-quirk for MSI MS-7350 motherboard. Add pci-quirk for MSI MS-7350 motherboard with Realtek ALC888. Signed-off-by: Sasha Alexandr Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d22b2606801..a2cebd7c135 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9069,6 +9069,7 @@ 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), -- cgit v1.2.3-18-g5258 From c259249f7d7a6e534f3bafe046929fee236eebfa Mon Sep 17 00:00:00 2001 From: Sasha Alexandr Date: Tue, 16 Jun 2009 14:52:54 -0400 Subject: ALSA: HDA - Name-fixes in code (tagra/targa) Correct some cut+paste typos from 'tagra' to 'targa'. Signed-off-by: Sasha Alexandr Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a2cebd7c135..dc77d75c355 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8068,7 +8068,7 @@ 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 +8088,7 @@ 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), @@ -8417,7 +8417,7 @@ 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 +8626,8 @@ 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) { @@ -9166,8 +9166,8 @@ 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, @@ -9175,12 +9175,12 @@ 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, @@ -9189,13 +9189,13 @@ 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), @@ -9207,8 +9207,8 @@ 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 }, @@ -9362,7 +9362,7 @@ 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, -- cgit v1.2.3-18-g5258 From def319f9e937f7a6a29718d3e2826c6c32f33245 Mon Sep 17 00:00:00 2001 From: Sasha Alexandr Date: Tue, 16 Jun 2009 16:00:15 -0400 Subject: ALSA: HDA - Correct trivial typos in comments. Correct some trivial typos in comments. Signed-off-by: Sasha Alexandr Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc77d75c355..8cebe265322 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -970,7 +970,7 @@ 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 +1170,7 @@ 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) @@ -6347,7 +6347,7 @@ 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 +7047,7 @@ 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 @@ -8957,7 +8957,7 @@ 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 @@ -11132,7 +11132,7 @@ 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 @@ -12287,7 +12287,7 @@ 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 @@ -13198,7 +13198,7 @@ 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 @@ -14060,7 +14060,7 @@ 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 @@ -14583,7 +14583,7 @@ 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 @@ -15028,7 +15028,7 @@ 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 @@ -15207,7 +15207,7 @@ 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) @@ -16670,7 +16670,7 @@ 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 -- cgit v1.2.3-18-g5258 From ef39412622b6e8f09c383de9565b07e93553fc27 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Jun 2009 11:42:45 +0200 Subject: ASoC: Kill BUS_ID_SIZE Remove the use of BUS_ID_SIZE from txx9aclc.c, as BUS_ID_SIZE will be removed soon later. Also, use snprintf() instead of sprintf() as a safer operation. Signed-off-by: Takashi Iwai --- sound/soc/txx9/txx9aclc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index fa336616152..938a58a5a24 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; -- cgit v1.2.3-18-g5258 From 0a842c8b60411e200b8a44b65dd78d9665692b91 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 17 Jun 2009 17:45:11 +0200 Subject: ALSA: snd_usb_caiaq: fix legacy input streaming Seems that nobody recently tried the input on the very first supported sound card model, RK2. This patch fixes the byte offset to make it running again. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 5 +++-- sound/usb/caiaq/device.c | 2 +- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b1445134216..8f9b60c5d74 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 22406245a98..0e5db719de2 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -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}," -- cgit v1.2.3-18-g5258 From d2fd4b09c07ae0c5ac288c0da6100c26ba9db15b Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Sun, 21 Jun 2009 00:40:10 +0100 Subject: ALSA: hda - Acer Inspire 6530G model for Realtek ALC888 The selected 4930G model seemed to keep the subwoofer 'tuba' function from operating correctly. Removing the existing PCI ID match made this work again, but it was mapped to 'Side' instead of to LFE as one would expect. This attempts to enable all functionality and keep the amount of available mixer sliders low. Any slider that had no audible effect on the output audio has been removed, and as such EAPD is not currently enabled. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 72 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8cebe265322..63cfebb8d95 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, @@ -1470,6 +1471,25 @@ 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" */ { @@ -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), @@ -9021,7 +9073,7 @@ 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 */ @@ -9256,6 +9308,24 @@ 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 }, -- cgit v1.2.3-18-g5258 From b1a914690c581f8f88b897d83a79b1c6eaf494c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 21 Jun 2009 10:56:44 +0200 Subject: ALSA: hda - Add model=6530g option Add the new model string corresponding to the previous Acer Aspire 6530G support. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index de8e10a9410..0d8d23581c4 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/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63cfebb8d95..bf4b78a74a8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9030,6 +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", -- cgit v1.2.3-18-g5258