From 4e01f54bfd3f423db8fd6c91c4f0471f18aa0c50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Apr 2009 08:53:34 +0200 Subject: ALSA: hda - Add Creative CA0110-IBG support Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode. In the HD-audio mode, no multiple streams are supported by just it behaves like a normal HD-audio device. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 13 + sound/pci/hda/Makefile | 4 + sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_intel.c | 5 + sound/pci/hda/patch_ca0110.c | 574 +++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 597 insertions(+) create mode 100644 sound/pci/hda/patch_ca0110.c (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index eb2a19b894a0..c710150d5065 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -139,6 +139,19 @@ config SND_HDA_CODEC_CONEXANT snd-hda-codec-conexant. This module is automatically loaded at probing. +config SND_HDA_CODEC_CA0110 + bool "Build Creative CA0110-IBG codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Creative CA0110-IBG codec support in + snd-hda-intel driver, found on some Creative X-Fi cards. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-ca0110. + This module is automatically loaded at probing. + config SND_HDA_CODEC_CMEDIA bool "Build C-Media HD-audio codec support" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 50f9d0967251..e3081d4586cc 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -13,6 +13,7 @@ snd-hda-codec-analog-objs := patch_analog.o snd-hda-codec-idt-objs := patch_sigmatel.o snd-hda-codec-si3054-objs := patch_si3054.o snd-hda-codec-atihdmi-objs := patch_atihdmi.o +snd-hda-codec-ca0110-objs := patch_ca0110.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o @@ -40,6 +41,9 @@ endif ifdef CONFIG_SND_HDA_CODEC_ATIHDMI obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o endif +ifdef CONFIG_SND_HDA_CODEC_CA0110 +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o +endif ifdef CONFIG_SND_HDA_CODEC_CONEXANT obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fd6e6f337d10..37f24ce7c3a2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -48,6 +48,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, { 0x11c1, "LSI" }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21e99cfa8c49..21a3092fad00 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2513,6 +2513,11 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, + /* Creative X-Fi (CA0110-IBG) */ + { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GENERIC }, /* AMD Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c new file mode 100644 index 000000000000..7ec41daa3f0c --- /dev/null +++ b/sound/pci/hda/patch_ca0110.c @@ -0,0 +1,574 @@ +/* + * HD audio interface patch for Creative X-Fi CA0110-IBG chip + * + * Copyright (c) 2008 Takashi Iwai + * + * This driver 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 driver 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 +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + +/* + */ + +struct ca0110_spec { + struct auto_pin_cfg autocfg; + struct hda_multi_out multiout; + hda_nid_t out_pins[AUTO_CFG_MAX_OUTS]; + hda_nid_t dacs[AUTO_CFG_MAX_OUTS]; + hda_nid_t hp_dac; + hda_nid_t input_pins[AUTO_PIN_LAST]; + hda_nid_t adcs[AUTO_PIN_LAST]; + hda_nid_t dig_out; + hda_nid_t dig_in; + unsigned int num_inputs; + const char *input_labels[AUTO_PIN_LAST]; + struct hda_pcm pcm_rec[2]; /* PCM information */ +}; + +/* + * PCM callbacks + */ +static int ca0110_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); +} + +static int ca0110_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + +static int ca0110_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); +} + +/* + * Digital out + */ +static int ca0110_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int ca0110_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int ca0110_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +/* + * Analog capture + */ +static int ca0110_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 ca0110_spec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); + return 0; +} + +static int ca0110_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + + snd_hda_codec_cleanup_stream(codec, spec->adcs[substream->number]); + return 0; +} + +/* + */ + +static char *dirstr[2] = { "Playback", "Capture" }; + +static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); +} + +static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); +} + +#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0) +#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0) +#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1) +#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1) +#define add_mono_switch(codec, nid, pfx, chan) \ + _add_switch(codec, nid, pfx, chan, 0) +#define add_mono_volume(codec, nid, pfx, chan) \ + _add_volume(codec, nid, pfx, chan, 0) + +static int ca0110_build_controls(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + static char *prefix[AUTO_CFG_MAX_OUTS] = { + "Front", "Surround", NULL, "Side", "Multi" + }; + hda_nid_t mutenid; + int i, err; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (get_wcaps(codec, spec->out_pins[i]) & AC_WCAP_OUT_AMP) + mutenid = spec->out_pins[i]; + else + mutenid = spec->multiout.dac_nids[i]; + if (!prefix[i]) { + err = add_mono_switch(codec, mutenid, + "Center", 1); + if (err < 0) + return err; + err = add_mono_switch(codec, mutenid, + "LFE", 1); + if (err < 0) + return err; + err = add_mono_volume(codec, spec->multiout.dac_nids[i], + "Center", 1); + if (err < 0) + return err; + err = add_mono_volume(codec, spec->multiout.dac_nids[i], + "LFE", 1); + if (err < 0) + return err; + } else { + err = add_out_switch(codec, mutenid, + prefix[i]); + if (err < 0) + return err; + err = add_out_volume(codec, spec->multiout.dac_nids[i], + prefix[i]); + if (err < 0) + return err; + } + } + if (cfg->hp_outs) { + if (get_wcaps(codec, cfg->hp_pins[0]) & AC_WCAP_OUT_AMP) + mutenid = cfg->hp_pins[0]; + else + mutenid = spec->multiout.dac_nids[i]; + + err = add_out_switch(codec, mutenid, "Headphone"); + if (err < 0) + return err; + if (spec->hp_dac) { + err = add_out_volume(codec, spec->hp_dac, "Headphone"); + if (err < 0) + return err; + } + } + for (i = 0; i < spec->num_inputs; i++) { + const char *label = spec->input_labels[i]; + if (get_wcaps(codec, spec->input_pins[i]) & AC_WCAP_IN_AMP) + mutenid = spec->input_pins[i]; + else + mutenid = spec->adcs[i]; + err = add_in_switch(codec, mutenid, label); + if (err < 0) + return err; + err = add_in_volume(codec, spec->adcs[i], label); + if (err < 0) + return err; + } + + if (spec->dig_out) { + err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out); + if (err < 0) + return err; + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; + } + if (spec->dig_in) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); + if (err < 0) + return err; + err = add_in_volume(codec, spec->dig_in, "IEC958"); + } + return 0; +} + +/* + */ +static struct hda_pcm_stream ca0110_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .ops = { + .open = ca0110_playback_pcm_open, + .prepare = ca0110_playback_pcm_prepare, + .cleanup = ca0110_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0110_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0110_capture_pcm_prepare, + .cleanup = ca0110_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0110_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .open = ca0110_dig_playback_pcm_open, + .close = ca0110_dig_playback_pcm_close, + .prepare = ca0110_dig_playback_pcm_prepare + }, +}; + +static struct hda_pcm_stream ca0110_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +static int ca0110_build_pcms(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->pcm_info = info; + codec->num_pcms = 0; + + info->name = "CA0110 Analog"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.num_dacs * 2; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; + codec->num_pcms++; + + if (!spec->dig_out && !spec->dig_in) + return 0; + + info++; + info->name = "CA0110 Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + ca0110_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out; + } + if (spec->dig_in) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0110_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; + } + codec->num_pcms++; + + return 0; +} + +static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + if (dac) + snd_hda_codec_write(codec, dac, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); +} + +static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } + if (adc) + snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); +} + +static int ca0110_init(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) + init_output(codec, spec->out_pins[i], + spec->multiout.dac_nids[i]); + init_output(codec, cfg->hp_pins[0], spec->hp_dac); + init_output(codec, cfg->dig_out_pins[0], spec->dig_out); + + for (i = 0; i < spec->num_inputs; i++) + init_input(codec, spec->input_pins[i], spec->adcs[i]); + init_input(codec, cfg->dig_in_pin, spec->dig_in); + return 0; +} + +static void ca0110_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +static struct hda_codec_ops ca0110_patch_ops = { + .build_controls = ca0110_build_controls, + .build_pcms = ca0110_build_pcms, + .init = ca0110_init, + .free = ca0110_free, +}; + + +static void parse_line_outs(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, n; + unsigned int def_conf; + hda_nid_t nid; + + n = 0; + for (i = 0; i < cfg->line_outs; i++) { + nid = cfg->line_out_pins[i]; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (!def_conf) + continue; /* invalid pin */ + if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1) + continue; + spec->out_pins[n++] = nid; + } + spec->multiout.dac_nids = spec->dacs; + spec->multiout.num_dacs = n; +} + +static void parse_hp_out(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + unsigned int def_conf; + hda_nid_t nid, dac; + + if (!cfg->hp_outs) + return; + nid = cfg->hp_pins[0]; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (!def_conf) { + cfg->hp_outs = 0; + return; + } + if (snd_hda_get_connections(codec, nid, &dac, 1) != 1) + return; + + for (i = 0; i < cfg->line_outs; i++) + if (dac == spec->dacs[i]) + break; + if (i >= cfg->line_outs) { + spec->hp_dac = dac; + spec->multiout.hp_nid = dac; + } +} + +static void parse_input(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid, pin; + int n, i, j; + + n = 0; + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int type = (wcaps & AC_WCAP_TYPE) >> + AC_WCAP_TYPE_SHIFT; + if (type != AC_WID_AUD_IN) + continue; + if (snd_hda_get_connections(codec, nid, &pin, 1) != 1) + continue; + if (pin == cfg->dig_in_pin) { + spec->dig_in = nid; + continue; + } + for (j = 0; j < AUTO_PIN_LAST; j++) + if (cfg->input_pins[j] == pin) + break; + if (j >= AUTO_PIN_LAST) + continue; + spec->input_pins[n] = pin; + spec->input_labels[n] = auto_pin_cfg_labels[j]; + spec->adcs[n] = nid; + n++; + } + spec->num_inputs = n; +} + +static void parse_digital(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (cfg->dig_outs && + snd_hda_get_connections(codec, cfg->dig_out_pins[0], + &spec->dig_out, 1) == 1) + spec->multiout.dig_out_nid = cfg->dig_out_pins[0]; +} + +static int ca0110_parse_auto_config(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + parse_line_outs(codec); + parse_hp_out(codec); + parse_digital(codec); + parse_input(codec); + return 0; +} + + +int patch_ca0110(struct hda_codec *codec) +{ + struct ca0110_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + codec->bus->needs_damn_long_delay = 1; + + err = ca0110_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = ca0110_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + + +/* + * patch entries + */ +static struct hda_codec_preset snd_hda_preset_ca0110[] = { + { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 }, + { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 }, + { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 }, + {} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:1102000a"); +MODULE_ALIAS("snd-hda-codec-id:1102000b"); +MODULE_ALIAS("snd-hda-codec-id:1102000d"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec"); + +static struct hda_codec_preset_list ca0110_list = { + .preset = snd_hda_preset_ca0110, + .owner = THIS_MODULE, +}; + +static int __init patch_ca0110_init(void) +{ + return snd_hda_add_codec_preset(&ca0110_list); +} + +static void __exit patch_ca0110_exit(void) +{ + snd_hda_delete_codec_preset(&ca0110_list); +} + +module_init(patch_ca0110_init) +module_exit(patch_ca0110_exit) -- cgit v1.2.3 From 18cb7109d3e83195b605ff2905981020e86f72ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Apr 2009 10:22:24 +0200 Subject: ALSA: hda - Check strcpy length Check the length to copy via strlen() beforehand to avoid the stack corruption, or use strlcpy() to be safe in HD-audio codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/hda_intel.c | 10 ++++++---- 2 files changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 37f24ce7c3a2..48f0cea7df14 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1431,6 +1431,8 @@ _snd_hda_find_mixer_ctl(struct hda_codec *codec, memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; id.index = idx; + if (snd_BUG_ON(strlen(name) >= sizeof(id.name))) + return NULL; strcpy(id.name, name); return snd_ctl_find_id(codec->bus->card, &id); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21a3092fad00..41db5d4da478 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1830,7 +1830,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, &pcm); if (err < 0) return err; - strcpy(pcm->name, cpcm->name); + strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); if (apcm == NULL) return -ENOMEM; @@ -2358,9 +2358,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } strcpy(card->driver, "HDA-Intel"); - strcpy(card->shortname, driver_short_names[chip->driver_type]); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->addr, chip->irq); + strlcpy(card->shortname, driver_short_names[chip->driver_type], + sizeof(card->shortname)); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); *rchip = chip; return 0; -- cgit v1.2.3 From 67667263674663767ddf4250bab2437a00ee780e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Apr 2009 10:49:25 +0200 Subject: ALSA: hda - Fix channels_max setting for CA0110 Added the missing definition of max channels for CA0110, which resulted in an error at opening PCM devices. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 7ec41daa3f0c..9398d92f18bd 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -309,7 +309,7 @@ static int ca0110_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.num_dacs * 2; + spec->multiout.max_channels; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; @@ -418,6 +418,7 @@ static void parse_line_outs(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = n; + spec->multiout.max_channels = n * 2; } static void parse_hp_out(struct hda_codec *codec) -- cgit v1.2.3 From 7670dc41b51983b369f9adfb8694a580e7b0cef2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Apr 2009 10:51:11 +0200 Subject: ALSA: hda - Use snd_hda_codec_get_pincfg() in patch_ca0110.c Use the new function to reduce the access and allow the user setup via sysfs, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 9398d92f18bd..392d108c3558 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -408,8 +408,7 @@ static void parse_line_outs(struct hda_codec *codec) n = 0; for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (!def_conf) continue; /* invalid pin */ if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1) @@ -432,8 +431,7 @@ static void parse_hp_out(struct hda_codec *codec) if (!cfg->hp_outs) return; nid = cfg->hp_pins[0]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (!def_conf) { cfg->hp_outs = 0; return; -- cgit v1.2.3 From 92c7c8a7d6e03eb4c0a3c5888e35dbc45f24744c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Mar 2009 07:32:14 +0100 Subject: ALSA: hda - Cache PCM and STREAM parameters queries Cache quries for PCM and STREAM parameters as well as ampcap and pincap sharing the hash table. This will reduce the superfluous access of the same codec verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 97 +++++++++++++++++++++++++++++------------------ 1 file changed, 61 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a4e5e5952115..3d8bf39e6d98 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1053,6 +1053,8 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* FIXME: more better hash key? */ #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) +#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) +#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) #define INFO_AMP_CAPS (1<<0) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) @@ -1143,19 +1145,32 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); -u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +static unsigned int +query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key, + unsigned int (*func)(struct hda_codec *, hda_nid_t)) { struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + info = get_alloc_amp_hash(codec, key); if (!info) return 0; if (!info->head.val) { - info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); info->head.val |= INFO_AMP_CAPS; + info->amp_caps = func(codec, nid); } return info->amp_caps; } + +static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); +} + +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), + read_pin_cap); +} EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); /* @@ -2538,6 +2553,41 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, } EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); +static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int val = 0; + if (nid != codec->afg && + (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) + val = snd_hda_param_read(codec, nid, AC_PAR_PCM); + if (!val || val == -1) + val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + if (!val || val == -1) + return 0; + return val; +} + +static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) +{ + return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid), + get_pcm_param); +} + +static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); + if (!streams || streams == -1) + streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); + if (!streams || streams == -1) + return 0; + return streams; +} + +static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) +{ + return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid), + get_stream_param); +} + /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats * @codec: the HDA codec @@ -2556,15 +2606,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, { unsigned int i, val, wcaps; - val = 0; wcaps = get_wcaps(codec, nid); - if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); - if (val == -1) - return -EIO; - } - if (!val) - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + val = query_pcm_param(codec, nid); if (ratesp) { u32 rates = 0; @@ -2586,15 +2629,9 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u64 formats = 0; unsigned int streams, bps; - streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); - if (streams == -1) + streams = query_stream_param(codec, nid); + if (!streams) return -EIO; - if (!streams) { - streams = snd_hda_param_read(codec, codec->afg, - AC_PAR_STREAM); - if (streams == -1) - return -EIO; - } bps = 0; if (streams & AC_SUPFMT_PCM) { @@ -2668,17 +2705,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, int i; unsigned int val = 0, rate, stream; - if (nid != codec->afg && - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); - if (val == -1) - return 0; - } - if (!val) { - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); - if (val == -1) - return 0; - } + val = query_pcm_param(codec, nid); + if (!val) + return 0; rate = format & 0xff00; for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++) @@ -2690,12 +2719,8 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, if (i >= AC_PAR_PCM_RATE_BITS) return 0; - stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); - if (stream == -1) - return 0; - if (!stream && nid != codec->afg) - stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); - if (!stream || stream == -1) + stream = query_stream_param(codec, nid); + if (!stream) return 0; if (stream & AC_SUPFMT_PCM) { -- cgit v1.2.3 From b613291fb21a2d74eb8323d97fe9aa5d281b306c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Mar 2009 07:36:09 +0100 Subject: ALSA: hda - Retry codec-verbs at errors The current error-recovery scheme for the codec communication errors doesn't work always well. Especially falling back to the single-command mode causes the fatal problem on many systems. In this patch, the problematic verb is re-issued again after the error (even with polling mode) instead of the single-cmd mode. The single-cmd mode will be used only when specified via the command option explicitly, mainly just for testing. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 17 +++++++++++++---- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 19 ++++++++++--------- 3 files changed, 24 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3d8bf39e6d98..1736ccbebc72 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -174,14 +174,23 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, unsigned int verb, unsigned int parm) { struct hda_bus *bus = codec->bus; - unsigned int res; + unsigned int cmd, res; + int repeated = 0; - res = make_codec_cmd(codec, nid, direct, verb, parm); + cmd = make_codec_cmd(codec, nid, direct, verb, parm); snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); - if (!bus->ops.command(bus, res)) + again: + if (!bus->ops.command(bus, cmd)) { res = bus->ops.get_response(bus); - else + if (res == -1 && bus->rirb_error) { + if (repeated++ < 1) { + snd_printd(KERN_WARNING "hda_codec: " + "Trying verb 0x%08x again\n", cmd); + goto again; + } + } + } else res = (unsigned int)-1; mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2fdecf4b0eb6..cd8979c7670b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -623,6 +623,7 @@ struct hda_bus { /* misc op flags */ unsigned int needs_damn_long_delay :1; unsigned int shutdown :1; /* being unloaded */ + unsigned int rirb_error:1; /* error in codec communication */ }; /* diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 30829ee920c3..803b72098ed3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -604,6 +604,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) } if (!chip->rirb.cmds) { smp_rmb(); + bus->rirb_error = 0; return chip->rirb.res; /* the last value */ } if (time_after(jiffies, timeout)) @@ -623,8 +624,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) chip->irq = -1; pci_disable_msi(chip->pci); chip->msi = 0; - if (azx_acquire_irq(chip, 1) < 0) + if (azx_acquire_irq(chip, 1) < 0) { + bus->rirb_error = 1; return -1; + } goto again; } @@ -644,14 +647,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) return -1; } - snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " - "switching to single_cmd mode: last cmd=0x%08x\n", - chip->last_cmd); - chip->rirb.rp = azx_readb(chip, RIRBWP); - chip->rirb.cmds = 0; - /* switch to single_cmd mode */ - chip->single_cmd = 1; - azx_free_cmd_io(chip); + snd_printk(KERN_ERR "hda_intel: azx_get_response timeout (ERROR): " + "last cmd=0x%08x\n", chip->last_cmd); + spin_lock_irq(&chip->reg_lock); + chip->rirb.cmds = 0; /* reset the index */ + bus->rirb_error = 1; + spin_unlock_irq(&chip->reg_lock); return -1; } -- cgit v1.2.3 From 586be3fcf97eec22fbc0ef6d67e823706aea7167 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Mar 2009 07:43:24 +0100 Subject: ALSA: hda - Add debug prints for Realtek auto-init Added a couple of debug prints to show the checked id numbers in alc_subsystem_id(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82097790f6f3..ee92c73df083 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1022,6 +1022,9 @@ static void alc_subsystem_id(struct hda_codec *codec, if (codec->vendor_id == 0x10ec0260) nid = 0x17; ass = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("realtek: No valid SSID, " + "checking pincfg 0x%08x for NID 0x%x\n", + nid, ass); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ @@ -1036,6 +1039,8 @@ static void alc_subsystem_id(struct hda_codec *codec, if (((ass >> 16) & 0xf) != tmp) return; do_sku: + snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", + ass & 0xffff, codec->vendor_id); /* * 0 : override * 1 : Swap Jack -- cgit v1.2.3 From a3b48c88f2d5a34c0e25aec0a3dab8069e5a9a72 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 13:37:29 +0200 Subject: ALSA: hda - minor optimization in hda_set_power_state() Check the target power-state before checking EAPD exception to reduce unneeded verb executions. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b649033a4c81..b91f6ed5cc58 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2348,7 +2348,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (wcaps & AC_WCAP_POWER) { unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wid_type == AC_WID_PIN) { + if (power_state == AC_PWRST_D3 && + wid_type == AC_WID_PIN) { unsigned int pincap; /* * don't power down the widget if it controls @@ -2360,7 +2361,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, nid, 0, AC_VERB_GET_EAPD_BTLENABLE, 0); eapd &= 0x02; - if (power_state == AC_PWRST_D3 && eapd) + if (eapd) continue; } } -- cgit v1.2.3 From dfed0ef9b3ff9e37903920b6938ed33344ad0b3d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 18:33:12 +0200 Subject: ALSA: hda - Fix a typo in debug print for realtek auto-detection The NID and ASS numbers were swapped... Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a6ec87a5c066..887712046c00 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1026,7 +1026,7 @@ static void alc_subsystem_id(struct hda_codec *codec, ass = snd_hda_codec_get_pincfg(codec, nid); snd_printd("realtek: No valid SSID, " "checking pincfg 0x%08x for NID 0x%x\n", - nid, ass); + ass, nid); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ -- cgit v1.2.3 From 4a79ba34cada6a5a4ee86ed53aa8a73ba1e6fc51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Apr 2009 16:31:35 +0200 Subject: ALSA: hda - Add amp initialization for realtek auto mode In the realtek auto-probing mode, the initialization of amp with some magic COEF or EAPD verbs is applied only when the codec SSID has valid values to satisfy the realtek's definition. However, many devices don't provide in that way, thus the device doesn't work as is. This patch allows the same initialization code even if the SSID doesn't pass the bit test. Also, alc_subsystem_id() is changed just to check and define the type, so that it's called in the parser, instead of the initializer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 239 +++++++++++++++++++++++++----------------- 1 file changed, 145 insertions(+), 94 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 583603f449b4..3a6306302c70 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -253,6 +253,15 @@ enum { /* for GPIO Poll */ #define GPIO_MASK 0x03 +/* extra amp-initialization sequence types */ +enum { + ALC_INIT_NONE, + ALC_INIT_DEFAULT, + ALC_INIT_GPIO1, + ALC_INIT_GPIO2, + ALC_INIT_GPIO3, +}; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -322,6 +331,7 @@ struct alc_spec { /* other flags */ unsigned int no_analog :1; /* digital I/O only */ + int init_amp; /* for virtual master */ hda_nid_t vmaster_nid; @@ -994,74 +1004,21 @@ static void alc888_coef_init(struct hda_codec *codec) AC_VERB_SET_PROC_COEF, 0x3030); } -/* 32-bit subsystem ID for BIOS loading in HD Audio codec. - * 31 ~ 16 : Manufacture ID - * 15 ~ 8 : SKU ID - * 7 ~ 0 : Assembly ID - * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36 - */ -static void alc_subsystem_id(struct hda_codec *codec, - unsigned int porta, unsigned int porte, - unsigned int portd) +static void alc_auto_init_amp(struct hda_codec *codec, int type) { - unsigned int ass, tmp, i; - unsigned nid; - struct alc_spec *spec = codec->spec; - - ass = codec->subsystem_id & 0xffff; - if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) - goto do_sku; - - /* - * 31~30 : port conetcivity - * 29~21 : reserve - * 20 : PCBEEP input - * 19~16 : Check sum (15:1) - * 15~1 : Custom - * 0 : override - */ - nid = 0x1d; - if (codec->vendor_id == 0x10ec0260) - nid = 0x17; - ass = snd_hda_codec_get_pincfg(codec, nid); - snd_printd("realtek: No valid SSID, " - "checking pincfg 0x%08x for NID 0x%x\n", - ass, nid); - if (!(ass & 1) && !(ass & 0x100000)) - return; - if ((ass >> 30) != 1) /* no physical connection */ - return; + unsigned int tmp; - /* check sum */ - tmp = 0; - for (i = 1; i < 16; i++) { - if ((ass >> i) & 1) - tmp++; - } - if (((ass >> 16) & 0xf) != tmp) - return; -do_sku: - snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", - ass & 0xffff, codec->vendor_id); - /* - * 0 : override - * 1 : Swap Jack - * 2 : 0 --> Desktop, 1 --> Laptop - * 3~5 : External Amplifier control - * 7~6 : Reserved - */ - tmp = (ass & 0x38) >> 3; /* external Amp control */ - switch (tmp) { - case 1: + switch (type) { + case ALC_INIT_GPIO1: snd_hda_sequence_write(codec, alc_gpio1_init_verbs); break; - case 3: + case ALC_INIT_GPIO2: snd_hda_sequence_write(codec, alc_gpio2_init_verbs); break; - case 7: + case ALC_INIT_GPIO3: snd_hda_sequence_write(codec, alc_gpio3_init_verbs); break; - case 5: /* set EAPD output high */ + case ALC_INIT_DEFAULT: switch (codec->vendor_id) { case 0x10ec0260: snd_hda_codec_write(codec, 0x0f, 0, @@ -1115,7 +1072,7 @@ do_sku: tmp | 0x2010); break; case 0x10ec0888: - /*alc888_coef_init(codec);*/ /* called in alc_init() */ + alc888_coef_init(codec); break; case 0x10ec0267: case 0x10ec0268: @@ -1130,7 +1087,104 @@ do_sku: tmp | 0x3000); break; } - default: + break; + } +} + +static void alc_init_auto_hp(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->autocfg.hp_pins[0]) + return; + + if (!spec->autocfg.speaker_pins[0]) { + if (spec->autocfg.line_out_pins[0]) + spec->autocfg.speaker_pins[0] = + spec->autocfg.line_out_pins[0]; + else + return; + } + + snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_HP_EVENT); + spec->unsol_event = alc_sku_unsol_event; +} + +/* check subsystem ID and set up device-specific initialization; + * return 1 if initialized, 0 if invalid SSID + */ +/* 32-bit subsystem ID for BIOS loading in HD Audio codec. + * 31 ~ 16 : Manufacture ID + * 15 ~ 8 : SKU ID + * 7 ~ 0 : Assembly ID + * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36 + */ +static int alc_subsystem_id(struct hda_codec *codec, + hda_nid_t porta, hda_nid_t porte, + hda_nid_t portd) +{ + unsigned int ass, tmp, i; + unsigned nid; + struct alc_spec *spec = codec->spec; + + ass = codec->subsystem_id & 0xffff; + if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) + goto do_sku; + + /* invalid SSID, check the special NID pin defcfg instead */ + /* + * 31~30 : port conetcivity + * 29~21 : reserve + * 20 : PCBEEP input + * 19~16 : Check sum (15:1) + * 15~1 : Custom + * 0 : override + */ + nid = 0x1d; + if (codec->vendor_id == 0x10ec0260) + nid = 0x17; + ass = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("realtek: No valid SSID, " + "checking pincfg 0x%08x for NID 0x%x\n", + nid, ass); + if (!(ass & 1) && !(ass & 0x100000)) + return 0; + if ((ass >> 30) != 1) /* no physical connection */ + return 0; + + /* check sum */ + tmp = 0; + for (i = 1; i < 16; i++) { + if ((ass >> i) & 1) + tmp++; + } + if (((ass >> 16) & 0xf) != tmp) + return 0; +do_sku: + snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", + ass & 0xffff, codec->vendor_id); + /* + * 0 : override + * 1 : Swap Jack + * 2 : 0 --> Desktop, 1 --> Laptop + * 3~5 : External Amplifier control + * 7~6 : Reserved + */ + tmp = (ass & 0x38) >> 3; /* external Amp control */ + switch (tmp) { + case 1: + spec->init_amp = ALC_INIT_GPIO1; + break; + case 3: + spec->init_amp = ALC_INIT_GPIO2; + break; + case 7: + spec->init_amp = ALC_INIT_GPIO3; + break; + case 5: + spec->init_amp = ALC_INIT_DEFAULT; break; } @@ -1138,7 +1192,7 @@ do_sku: * when the external headphone out jack is plugged" */ if (!(ass & 0x8000)) - return; + return 1; /* * 10~8 : Jack location * 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered @@ -1146,14 +1200,6 @@ do_sku: * 15 : 1 --> enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ - if (!spec->autocfg.speaker_pins[0]) { - if (spec->autocfg.line_out_pins[0]) - spec->autocfg.speaker_pins[0] = - spec->autocfg.line_out_pins[0]; - else - return; - } - if (!spec->autocfg.hp_pins[0]) { tmp = (ass >> 11) & 0x3; /* HP to chassis */ if (tmp == 0) @@ -1163,23 +1209,23 @@ do_sku: else if (tmp == 2) spec->autocfg.hp_pins[0] = portd; else - return; + return 1; } - if (spec->autocfg.hp_pins[0]) - snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC880_HP_EVENT); -#if 0 /* it's broken in some acses -- temporarily disabled */ - if (spec->autocfg.input_pins[AUTO_PIN_MIC] && - spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]) - snd_hda_codec_write(codec, - spec->autocfg.input_pins[AUTO_PIN_MIC], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC880_MIC_EVENT); -#endif /* disabled */ + alc_init_auto_hp(codec); + return 1; +} - spec->unsol_event = alc_sku_unsol_event; +static void alc_ssid_check(struct hda_codec *codec, + hda_nid_t porta, hda_nid_t porte, hda_nid_t portd) +{ + if (!alc_subsystem_id(codec, porta, porte, portd)) { + struct alc_spec *spec = codec->spec; + snd_printd("realtek: " + "Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + alc_init_auto_hp(codec); + } } /* @@ -2923,8 +2969,7 @@ static int alc_init(struct hda_codec *codec) unsigned int i; alc_fix_pll(codec); - if (codec->vendor_id == 0x10ec0888) - alc888_coef_init(codec); + alc_auto_init_amp(codec, spec->init_amp); for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); @@ -4198,7 +4243,6 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i < spec->autocfg.line_outs; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -4303,6 +4347,8 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -5678,7 +5724,6 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t nid; - alc_subsystem_id(codec, 0x10, 0x15, 0x0f); nid = spec->autocfg.line_out_pins[0]; if (nid) { int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -5788,6 +5833,8 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + alc_ssid_check(codec, 0x10, 0x15, 0x0f); + return 1; } @@ -7013,7 +7060,6 @@ static void alc882_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -9154,7 +9200,6 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -9317,6 +9362,7 @@ static int patch_alc883(struct hda_codec *codec) if (!spec->capsrc_nids) spec->capsrc_nids = alc883_capsrc_nids; spec->capture_style = CAPT_MIX; /* matrix-style capture */ + spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; @@ -10842,6 +10888,8 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x14, 0x1b); + return 1; } @@ -13925,7 +13973,6 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x0e, 0x0f, 0x0b); for (i = 0; i < spec->autocfg.line_outs; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -14008,6 +14055,8 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(spec); + alc_ssid_check(codec, 0x0e, 0x0f, 0x0b); + return 1; } @@ -14889,7 +14938,6 @@ static void alc861vd_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -15107,6 +15155,8 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -16931,7 +16981,6 @@ static void alc662_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -17028,6 +17077,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } -- cgit v1.2.3 From 2a2ed0dfc9ec44a899c7d4672f73f2c045099118 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 13:01:26 +0200 Subject: ALSA: hda - Don't enable auto-mute but for speakers in patch_realtek.c Enable auto-muting in model=auto only for devices with HP and speakers. For devices with HP and line-outs, don't enable the auto-muting. Also, add a debug print to show the auto-mute feature. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a6306302c70..96475dc95fbb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1099,13 +1099,16 @@ static void alc_init_auto_hp(struct hda_codec *codec) return; if (!spec->autocfg.speaker_pins[0]) { - if (spec->autocfg.line_out_pins[0]) + if (spec->autocfg.line_out_pins[0] && + spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) spec->autocfg.speaker_pins[0] = spec->autocfg.line_out_pins[0]; else return; } + snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", + spec->autocfg.hp_pins[0]); snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); -- cgit v1.2.3 From cb6605c1e4d2a2eaffdde433fbfe1567ca688458 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 13:03:19 +0200 Subject: ALSA: hda - Fix a typo in patch_realtek.c again The commmit dfed0ef9b3ff9e37903920b6938ed33344ad0b3d was reverted accidentally by the merge of auto-detection fix patch. Fixed again now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96475dc95fbb..3e7207b927c8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1151,7 +1151,7 @@ static int alc_subsystem_id(struct hda_codec *codec, ass = snd_hda_codec_get_pincfg(codec, nid); snd_printd("realtek: No valid SSID, " "checking pincfg 0x%08x for NID 0x%x\n", - nid, ass); + ass, nid); if (!(ass & 1) && !(ass & 0x100000)) return 0; if ((ass >> 30) != 1) /* no physical connection */ -- cgit v1.2.3 From 514bf54cd8c7f172816d3c003a6d022e9165a29b Mon Sep 17 00:00:00 2001 From: James Gardiner Date: Sun, 3 May 2009 04:00:44 -0400 Subject: ALSA: hda - Addition for HP dv4-1222nr laptop support Signed-off-by: James Gardiner Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 44 ++++++++++++++++++++++++---- 2 files changed, 39 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8eec05bc079e..36c97126d17d 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -347,6 +347,7 @@ STAC92HD71B* hp-m4 HP mini 1000 hp-dv5 HP dv series hp-hdx HP HDX series + hp-dv4-1222nr HP dv4-1222nr (with LED support) auto BIOS setup (default) STAC92HD73* diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 917bc5d3ac2c..76487de33c84 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1836,6 +1837,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = NULL, [STAC_HP_DV5] = NULL, [STAC_HP_HDX] = NULL, + [STAC_HP_DV4_1222NR] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { @@ -1847,6 +1849,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -1855,6 +1858,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, + "HP dv4-1222nr", STAC_HP_DV4_1222NR), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -4520,27 +4525,38 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } - /* - * using power check for controlling mute led of HP HDX notebooks + * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) * * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise * the LED is NOT working properly ! + * + * Changed name to reflect that it now works for any designated + * model, not just HP HDX. */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, +static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_bit = 0; /* gets rid of compiler warning */ + + switch (spec->board_config) { + case STAC_HP_DV4_1222NR: + gpio_bit = 0x01; + break; + case STAC_HP_HDX: + gpio_bit = 0x08; + } if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data &= ~0x08; /* orange */ + spec->gpio_data &= ~gpio_bit; /* orange */ else - spec->gpio_data |= 0x08; /* white */ + spec->gpio_data |= gpio_bit; /* white */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, @@ -5219,6 +5235,22 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: + spec->num_dmics = 1; + /* I don't know if it needs 1 or 2 smuxes - will wait for + * bug reports to fix if needed + */ + spec->num_smuxes = 1; + spec->num_dmuxes = 1; +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* This controls MUTE LED */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); @@ -5239,7 +5271,7 @@ again: /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_hdx_check_power_status; + stac92xx_hp_check_power_status; #endif break; }; -- cgit v1.2.3 From 41d5545d23d2ccf4d34725094dccebd37f15c1c4 Mon Sep 17 00:00:00 2001 From: Kacper Szczesniak Date: Thu, 7 May 2009 12:47:43 +0200 Subject: ALSA: hda - Add support for MacBook 5.1 (Aluminium) Signed-off-by: Kacper Szczesniak Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 73 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3e7207b927c8..b9495342c921 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -205,6 +205,7 @@ enum { ALC882_ASUS_A7M, ALC885_MACPRO, ALC885_MBP3, + ALC885_MB5, ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, @@ -6164,6 +6165,16 @@ static struct hda_input_mux alc882_capture_source = { { "CD", 0x4 }, }, }; + +static struct hda_input_mux mb5_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + /* * 2ch mode */ @@ -6293,6 +6304,20 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), { } /* end */ }; + +static struct snd_kcontrol_new alc885_mb5_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 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("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), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT), + { } /* end */ +}; static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6520,6 +6545,38 @@ static struct hda_verb alc882_macpro_init_verbs[] = { { } }; +/* Macbook 5,1 */ +static struct hda_verb alc885_mb5_init_verbs[] = { + /* Front mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LineOut mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin: output 0 (0x0d) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* HP Pin: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin */ + {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)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -6864,6 +6921,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7J] = "asus-a7j", [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", + [ALC885_MB5] = "mb5", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", @@ -6944,6 +7002,18 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mbp3_unsol_event, .init_hook = alc885_mbp3_automute, }, + [ALC885_MB5] = { + .mixers = { alc885_mb5_mixer }, + .init_verbs = { alc885_mb5_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mbp_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .input_mux = &mb5_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, @@ -7249,6 +7319,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; + case 0x106b3f00: /* Macbook 5,1 */ + board_config = ALC885_MB5; + break; default: /* ALC889A is handled better as ALC888-compatible */ if (codec->revision_id == 0x100101 || -- cgit v1.2.3 From 9da29271bea5d831d745f3ceb7f6f6b2def13a5b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 May 2009 16:31:14 +0200 Subject: ALSA: hda - Fix secondary SPDIF on VT1708S and VT1702 codecs VIA VT1708S and VT1702 codecs can have two SPDIF outputs. One of them should have been handled as the extra digital out, but it's not properly accessed. This patch fixes the handling of the secondary SPDIF on these codecs with the slave dig-out as found in patch_sigmatel.c. This makes the use of such a device easier (for normal users). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 111 +++++++++++++++++++++------------------------- 1 file changed, 51 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b25a5cc637d6..8e004fb6961a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -205,7 +205,7 @@ struct via_spec { /* playback */ struct hda_multi_out multiout; - hda_nid_t extra_dig_out_nid; + hda_nid_t slave_dig_outs[2]; /* capture */ unsigned int num_adc_nids; @@ -731,21 +731,6 @@ static int via_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_dig_close(codec, &spec->multiout); } -/* setup SPDIF output stream */ -static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid, - unsigned int stream_tag, unsigned int format) -{ - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_ctls & AC_DIG1_ENABLE) - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); - /* turn on again (if needed) */ - if (codec->spdif_ctls & AC_DIG1_ENABLE) - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); -} - static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -753,19 +738,16 @@ static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - hda_nid_t nid; - - /* 1st or 2nd S/PDIF */ - if (substream->number == 0) - nid = spec->multiout.dig_out_nid; - else if (substream->number == 1) - nid = spec->extra_dig_out_nid; - else - return -1; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} - mutex_lock(&codec->spdif_mutex); - setup_dig_playback_stream(codec, nid, stream_tag, format); - mutex_unlock(&codec->spdif_mutex); +static int via_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct via_spec *spec = codec->spec; + snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); return 0; } @@ -842,7 +824,8 @@ static struct hda_pcm_stream vt1708_pcm_digital_playback = { .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -874,13 +857,6 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; spec->multiout.share_spdif = 1; - - if (spec->extra_dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->extra_dig_out_nid); - if (err < 0) - return err; - } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -1013,10 +989,6 @@ static void via_unsol_event(struct hda_codec *codec, via_gpio_control(codec); } -static hda_nid_t slave_dig_outs[] = { - 0, -}; - static int via_init(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1051,8 +1023,9 @@ static int via_init(struct hda_codec *codec) snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - /* no slave outs */ - codec->slave_dig_outs = slave_dig_outs; + /* assign slave outs */ + if (spec->slave_dig_outs[0]) + codec->slave_dig_outs = spec->slave_dig_outs; return 0; } @@ -2134,7 +2107,8 @@ static struct hda_pcm_stream vt1708B_pcm_digital_playback = { .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -2589,14 +2563,15 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = { }; static struct hda_pcm_stream vt1708S_pcm_digital_playback = { - .substreams = 2, + .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -2805,14 +2780,37 @@ static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, return 0; } +/* fill out digital output widgets; one for master and one for slave outputs */ +static void fill_dig_outs(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->autocfg.dig_outs; i++) { + hda_nid_t nid; + int conn; + + nid = spec->autocfg.dig_out_pins[i]; + if (!nid) + continue; + conn = snd_hda_get_connections(codec, nid, &nid, 1); + if (conn < 1) + continue; + if (!spec->multiout.dig_out_nid) + spec->multiout.dig_out_nid = nid; + else { + spec->slave_dig_outs[0] = nid; + break; /* at most two dig outs */ + } + } +} + static int vt1708S_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int err; - static hda_nid_t vt1708s_ignore[] = {0x21, 0}; - err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, - vt1708s_ignore); + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); if (err < 0) return err; err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg); @@ -2833,10 +2831,7 @@ static int vt1708S_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 = VT1708S_DIGOUT_NID; - - spec->extra_dig_out_nid = 0x15; + fill_dig_outs(codec); if (spec->kctls.list) spec->mixers[spec->num_mixers++] = spec->kctls.list; @@ -3000,7 +2995,8 @@ static struct hda_pcm_stream vt1702_pcm_digital_playback = { .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -3128,10 +3124,8 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int err; - static hda_nid_t vt1702_ignore[] = {0x1C, 0}; - err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, - vt1702_ignore); + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); if (err < 0) return err; err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg); @@ -3152,10 +3146,7 @@ static int vt1702_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 = VT1702_DIGOUT_NID; - - spec->extra_dig_out_nid = 0x1B; + fill_dig_outs(codec); if (spec->kctls.list) spec->mixers[spec->num_mixers++] = spec->kctls.list; -- cgit v1.2.3 From 42171c17f267d7fdc5167ad7b6b5fb9edfd04186 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 May 2009 14:11:43 +0200 Subject: ALSA: hda - Fix and clean up hippo-compat HP auto-muting The speaker auto-muting per HP plugging for ALC262 HIPPO and compatible devices is slightly buggy as the "Master" or "Front" mixer control can still toggle the speaker output even if the headphone is plugged. This patch fixes the issue, and clean up the hippo-related codes together with fixes of some inconsistent mixer names. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 372 ++++++++++++++++++------------------------ 1 file changed, 158 insertions(+), 214 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b9495342c921..8b242c9da4dd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9534,24 +9534,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_hippo1_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - { } /* end */ -}; - /* update HP, line and mono-out pins according to the master switch */ static void alc262_hp_master_update(struct hda_codec *codec) { @@ -9772,46 +9754,132 @@ static struct hda_input_mux alc262_hp_rp5700_capture_source = { }, }; -/* bind hp and internal speaker mute (with plug check) */ -static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* bind hp and internal speaker mute (with plug check) as master switch */ +static void alc262_hippo_master_update(struct hda_codec *codec) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; + struct alc_spec *spec = codec->spec; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; + hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0]; + unsigned int mute; - /* change hp mute */ - change = snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); - if (change) { - /* change speaker according to HP jack state */ - struct alc_spec *spec = codec->spec; - unsigned int mute; - if (spec->jack_present) - mute = HDA_AMP_MUTE; - else - mute = snd_hda_codec_amp_read(codec, 0x15, 0, - HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + /* HP */ + mute = spec->master_sw ? 0 : HDA_AMP_MUTE; + snd_hda_codec_amp_stereo(codec, hp_nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + /* mute internal speaker per jack sense */ + if (spec->jack_present) + mute = HDA_AMP_MUTE; + if (line_nid) + snd_hda_codec_amp_stereo(codec, line_nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + if (speaker_nid && speaker_nid != line_nid) + snd_hda_codec_amp_stereo(codec, speaker_nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, mute); +} + +#define alc262_hippo_master_sw_get alc262_hp_master_sw_get + +static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int val = !!*ucontrol->value.integer.value; + + if (val == spec->master_sw) + return 0; + spec->master_sw = val; + alc262_hippo_master_update(codec); + return 1; +} + +#define ALC262_HIPPO_MASTER_SWITCH \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Master Playback Switch", \ + .info = snd_ctl_boolean_mono_info, \ + .get = alc262_hippo_master_sw_get, \ + .put = alc262_hippo_master_sw_put, \ } - return change; + +static struct snd_kcontrol_new alc262_hippo_mixer[] = { + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc262_hippo1_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_hippo_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + unsigned int present; + + /* need to execute and sync at first */ + snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, hp_nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + alc262_hippo_master_update(codec); } +static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_hippo_automute(codec); +} + +static void alc262_hippo_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc262_hippo_automute(codec); +} + +static void alc262_hippo1_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc262_hippo_automute(codec); +} + + static struct snd_kcontrol_new alc262_sony_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc262_sony_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - }, + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -9820,8 +9888,8 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { }; static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -10076,69 +10144,6 @@ static void alc262_toshiba_s06_init_hook(struct hda_codec *codec) alc262_dmic_automute(codec); } -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hippo_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - unsigned int mute; - unsigned int present; - - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc262_hippo_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hippo_automute(codec); -} - -static void alc262_hippo1_automute(struct hda_codec *codec) -{ - unsigned int mute; - unsigned int present; - - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc262_hippo1_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hippo1_automute(codec); -} - /* * nec model * 0x15 = headphone @@ -10406,14 +10411,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc262_sony_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - }, + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -11068,7 +11066,7 @@ static struct alc_config_preset alc262_presets[] = { .input_mux = &alc262_capture_source, }, [ALC262_HIPPO] = { - .mixers = { alc262_base_mixer }, + .mixers = { alc262_hippo_mixer }, .init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs}, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, @@ -11078,7 +11076,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_HIPPO_1] = { .mixers = { alc262_hippo1_mixer }, @@ -11090,8 +11088,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo1_unsol_event, - .init_hook = alc262_hippo1_automute, + .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo1_init_hook, }, [ALC262_FUJITSU] = { .mixers = { alc262_fujitsu_mixer }, @@ -11185,7 +11183,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, @@ -11197,7 +11195,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, @@ -11262,7 +11260,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_TYAN] = { .mixers = { alc262_tyan_mixer }, @@ -11419,6 +11417,17 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { { } }; +static struct snd_kcontrol_new alc268_toshiba_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + { } +}; + /* bind Beep switches of both NID 0x0f and 0x10 */ static struct hda_bind_ctls alc268_bind_beep_sw = { .ops = &snd_hda_bind_sw, @@ -11442,8 +11451,6 @@ static struct hda_verb alc268_eapd_verbs[] = { }; /* Toshiba specific */ -#define alc268_toshiba_automute alc262_hippo_automute - static struct hda_verb alc268_toshiba_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } /* end */ @@ -11579,13 +11586,8 @@ static struct hda_verb alc268_acer_verbs[] = { }; /* unsolicited event for HP jack sensing */ -static void alc268_toshiba_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc268_toshiba_automute(codec); -} +#define alc268_toshiba_unsol_event alc262_hippo_unsol_event +#define alc268_toshiba_init_hook alc262_hippo_init_hook static void alc268_acer_unsol_event(struct hda_codec *codec, unsigned int res) @@ -12230,7 +12232,7 @@ static struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, @@ -12244,7 +12246,7 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc268_toshiba_unsol_event, - .init_hook = alc268_toshiba_automute, + .init_hook = alc268_toshiba_init_hook, }, [ALC268_ACER] = { .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, @@ -12327,7 +12329,7 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc268_toshiba_unsol_event, - .init_hook = alc268_toshiba_automute + .init_hook = alc268_toshiba_init_hook }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { @@ -15552,10 +15554,8 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { }; static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -15568,15 +15568,11 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { }; static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), @@ -16084,51 +16080,25 @@ static void alc662_eeepc_mic_automute(struct hda_codec *codec) static void alc662_eeepc_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc262_hippo1_automute( codec ); - if ((res >> 26) == ALC880_MIC_EVENT) alc662_eeepc_mic_automute(codec); + else + alc262_hippo_unsol_event(codec, res); } static void alc662_eeepc_inithook(struct hda_codec *codec) { - alc262_hippo1_automute( codec ); + alc262_hippo1_init_hook(codec); alc662_eeepc_mic_automute(codec); } -static void alc662_eeepc_ep20_automute(struct hda_codec *codec) -{ - unsigned int mute; - unsigned int present; - - snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc662_eeepc_ep20_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc662_eeepc_ep20_automute(codec); -} - static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) { - alc662_eeepc_ep20_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc262_hippo_master_update(codec); } static void alc663_m51va_speaker_automute(struct hda_codec *codec) @@ -16462,35 +16432,9 @@ static void alc663_g50v_inithook(struct hda_codec *codec) alc662_eeepc_mic_automute(codec); } -/* bind hp and internal speaker mute (with plug check) */ -static int alc662_ecs_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); - if (change) - alc262_hippo1_automute(codec); - return change; -} - static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc662_ecs_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - }, + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -16682,7 +16626,7 @@ static struct alc_config_preset alc662_presets[] = { .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, - .unsol_event = alc662_eeepc_ep20_unsol_event, + .unsol_event = alc662_eeepc_unsol_event, .init_hook = alc662_eeepc_ep20_inithook, }, [ALC662_ECS] = { -- cgit v1.2.3 From b72519b518211ecb7c4c7b5e660ac6235ca7d1b7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 May 2009 14:31:55 +0200 Subject: ALSA: hda - Clean up for ALC262 HP model auto-mute functions Just clean up, no functional changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 54 +++++++++++++++---------------------------- 1 file changed, 19 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b242c9da4dd..7c043b3b81a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9589,14 +9589,7 @@ static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec, alc262_hp_wildwest_automute(codec); } -static int alc262_hp_master_sw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - *ucontrol->value.integer.value = spec->master_sw; - return 0; -} +#define alc262_hp_master_sw_get alc260_hp_master_sw_get static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -9612,14 +9605,17 @@ static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } +#define ALC262_HP_MASTER_SWITCH \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Master Playback Switch", \ + .info = snd_ctl_boolean_mono_info, \ + .get = alc262_hp_master_sw_get, \ + .put = alc262_hp_master_sw_put, \ + } + static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = alc262_hp_master_sw_get, - .put = alc262_hp_master_sw_put, - }, + ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -9643,13 +9639,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { }; static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = alc262_hp_master_sw_get, - .put = alc262_hp_master_sw_put, - }, + ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9676,17 +9666,14 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hp_t5735_automute(struct hda_codec *codec, int force) +static void alc262_hp_t5735_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + unsigned int present; - if (force || !spec->sense_updated) { - unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; - spec->sense_updated = 1; - } + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE, spec->jack_present ? HDA_AMP_MUTE : 0); } @@ -9696,13 +9683,10 @@ static void alc262_hp_t5735_unsol_event(struct hda_codec *codec, { if ((res >> 26) != ALC880_HP_EVENT) return; - alc262_hp_t5735_automute(codec, 1); + alc262_hp_t5735_automute(codec); } -static void alc262_hp_t5735_init_hook(struct hda_codec *codec) -{ - alc262_hp_t5735_automute(codec, 1); -} +#define alc262_hp_t5735_init_hook alc262_hp_t5735_automute static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), -- cgit v1.2.3 From a9fd4f3fcdc7e5757424f7e5888f660cf34bb1a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 May 2009 15:57:59 +0200 Subject: ALSA: hda - Clean up Realtek auto-mute unsol routines Most of unsol handlers defined in patch_realtek.c can be classified to two types, mute via amp of pins and mute via ctl bits of pins. Thus there are a big room to generalize each implementation. This patch creates two generic functions, alc_automute_amp() and alc_automute_pin(). The latter is actually changed from the previous alc_sku_automute(). Each caller needs to initialize hp_pins and speaker_pins properly at own init_hook. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 901 +++++++++++++++--------------------------- 1 file changed, 311 insertions(+), 590 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c043b3b81a5..34f6fb72f2bc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -926,20 +926,26 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, alc_fix_pll(codec); } -static void alc_sku_automute(struct hda_codec *codec) +static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int present; - unsigned int hp_nid = spec->autocfg.hp_pins[0]; - unsigned int sp_nid = spec->autocfg.speaker_pins[0]; + unsigned int nid = spec->autocfg.hp_pins[0]; + int i; /* need to execute and sync at first */ - snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, hp_nid, 0, + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->jack_present ? 0 : PIN_OUT); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + nid = spec->autocfg.speaker_pins[i]; + if (!nid) + break; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->jack_present ? 0 : PIN_OUT); + } } #if 0 /* it's broken in some acses -- temporarily disabled */ @@ -974,16 +980,19 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 28; else res >>= 26; - if (res == ALC880_HP_EVENT) - alc_sku_automute(codec); - - if (res == ALC880_MIC_EVENT) + switch (res) { + case ALC880_HP_EVENT: + alc_automute_pin(codec); + break; + case ALC880_MIC_EVENT: alc_mic_automute(codec); + break; + } } static void alc_inithook(struct hda_codec *codec) { - alc_sku_automute(codec); + alc_automute_pin(codec); alc_mic_automute(codec); } @@ -1364,32 +1373,58 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { {} }; -static void alc888_fujitsu_xa3530_automute(struct hda_codec *codec) +static void alc_automute_amp(struct hda_codec *codec) { - unsigned int present; - unsigned int bits; - /* Line out presence */ - present = snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - /* HP out presence */ - present = present || snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; + struct alc_spec *spec = codec->spec; + unsigned int val, mute; + hda_nid_t nid; + int i; + + spec->jack_present = 0; + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + nid = spec->autocfg.hp_pins[i]; + if (!nid) + break; + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + if (val & AC_PINSENSE_PRESENCE) { + spec->jack_present = 1; + break; + } + } + + mute = spec->jack_present ? HDA_AMP_MUTE : 0; /* Toggle internal speakers muting */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - /* Toggle internal bass muting */ - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + nid = spec->autocfg.speaker_pins[i]; + if (!nid) + break; + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } } -static void alc888_fujitsu_xa3530_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc_automute_amp_unsol_event(struct hda_codec *codec, + unsigned int res) { - if (res >> 26 == ALC880_HP_EVENT) - alc888_fujitsu_xa3530_automute(codec); + if (codec->vendor_id == 0x10ec0880) + res >>= 28; + else + res >>= 26; + if (res == ALC880_HP_EVENT) + alc_automute_amp(codec); } +static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x17; /* line-out */ + spec->autocfg.hp_pins[1] = 0x1b; /* hp */ + spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ + spec->autocfg.speaker_pins[1] = 0x15; /* bass */ + alc_automute_amp(codec); +} /* * ALC888 Acer Aspire 4930G model @@ -1456,22 +1491,13 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; -static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) +static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned int bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if (res >> 26 == ALC880_HP_EVENT) - alc888_acer_aspire_4930g_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } /* @@ -2439,21 +2465,6 @@ static struct hda_verb alc880_beep_init_verbs[] = { { } }; -/* toggle speaker-output according to the hp-jack state */ -static void alc880_uniwill_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - /* auto-toggle front mic */ static void alc880_uniwill_mic_automute(struct hda_codec *codec) { @@ -2466,9 +2477,14 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec) snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } -static void alc880_uniwill_automute(struct hda_codec *codec) +static void alc880_uniwill_init_hook(struct hda_codec *codec) { - alc880_uniwill_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x16; + alc_automute_amp(codec); alc880_uniwill_mic_automute(codec); } @@ -2479,24 +2495,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, * definition. 4bit tag is placed at 28 bit! */ switch (res >> 28) { - case ALC880_HP_EVENT: - alc880_uniwill_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc880_uniwill_mic_automute(codec); break; + default: + alc_automute_amp_unsol_event(codec, res); + break; } } -static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) +static void alc880_uniwill_p53_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + struct alc_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -2518,10 +2532,10 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - if ((res >> 28) == ALC880_HP_EVENT) - alc880_uniwill_p53_hp_automute(codec); if ((res >> 28) == ALC880_DCVOL_EVENT) alc880_uniwill_p53_dcvol_automute(codec); + else + alc_automute_amp_unsol_event(codec, res); } /* @@ -2753,30 +2767,18 @@ static struct hda_verb alc880_lg_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* jack sense */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } }; /* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_automute(struct hda_codec *codec) +static void alc880_lg_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - if ((res >> 28) == 0x01) - alc880_lg_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x17; + alc_automute_amp(codec); } /* @@ -2850,30 +2852,18 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* jack sense */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } }; /* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_lw_automute(struct hda_codec *codec) +static void alc880_lg_lw_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - if ((res >> 28) == 0x01) - alc880_lg_lw_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -2920,16 +2910,10 @@ static struct hda_verb alc880_medion_rim_init_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc880_medion_rim_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - if (present) + struct alc_spec *spec = codec->spec; + alc_automute_amp(codec); + /* toggle EAPD */ + if (spec->jack_present) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); else snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); @@ -2945,6 +2929,15 @@ static void alc880_medion_rim_unsol_event(struct hda_codec *codec, alc880_medion_rim_automute(codec); } +static void alc880_medion_rim_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc880_medion_rim_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE static struct hda_amp_list alc880_loopbacks[] = { { 0x0b, HDA_INPUT, 0 }, @@ -3803,7 +3796,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_f1734_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_hp_automute, + .init_hook = alc880_uniwill_p53_init_hook, }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, @@ -3880,7 +3873,7 @@ static struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_unsol_event, - .init_hook = alc880_uniwill_automute, + .init_hook = alc880_uniwill_init_hook, }, [ALC880_UNIWILL_P53] = { .mixers = { alc880_uniwill_p53_mixer }, @@ -3892,7 +3885,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_hp_automute, + .init_hook = alc880_uniwill_p53_init_hook, }, [ALC880_FUJITSU] = { .mixers = { alc880_fujitsu_mixer }, @@ -3906,7 +3899,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_hp_automute, + .init_hook = alc880_uniwill_p53_init_hook, }, [ALC880_CLEVO] = { .mixers = { alc880_three_stack_mixer }, @@ -3931,8 +3924,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_ch_modes, .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, - .unsol_event = alc880_lg_unsol_event, - .init_hook = alc880_lg_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc880_lg_init_hook, #ifdef CONFIG_SND_HDA_POWER_SAVE .loopbacks = alc880_lg_loopbacks, #endif @@ -3947,8 +3940,8 @@ static struct alc_config_preset alc880_presets[] = { .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes), .channel_mode = alc880_lg_lw_modes, .input_mux = &alc880_lg_lw_capture_source, - .unsol_event = alc880_lg_lw_unsol_event, - .init_hook = alc880_lg_lw_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc880_lg_lw_init_hook, }, [ALC880_MEDION_RIM] = { .mixers = { alc880_medion_rim_mixer }, @@ -3962,7 +3955,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_medion_rim_capture_source, .unsol_event = alc880_medion_rim_unsol_event, - .init_hook = alc880_medion_rim_automute, + .init_hook = alc880_medion_rim_init_hook, }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { @@ -6666,45 +6659,23 @@ static struct hda_verb alc885_imac24_init_verbs[] = { }; /* Toggle speaker-output according to the hp-jack state */ -static void alc885_imac24_automute(struct hda_codec *codec) +static void alc885_imac24_automute_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -/* Processes unsolicited events. */ -static void alc885_imac24_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_imac24_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + spec->autocfg.speaker_pins[1] = 0x1a; + alc_automute_amp(codec); } -static void alc885_mbp3_automute(struct hda_codec *codec) +static void alc885_mbp3_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE); + struct alc_spec *spec = codec->spec; -} -static void alc885_mbp3_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_mbp3_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } @@ -6729,24 +6700,25 @@ static struct hda_verb alc882_targa_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc882_targa_automute(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + struct alc_spec *spec = codec->spec; + alc_automute_amp(codec); snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - present ? 1 : 3); + spec->jack_present ? 1 : 3); +} + +static void alc882_targa_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc882_targa_automute(codec); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) { - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 26 bit! - */ - if (((res >> 26) == ALC880_HP_EVENT)) { + if ((res >> 26) == ALC880_HP_EVENT) alc882_targa_automute(codec); - } } static struct hda_verb alc882_asus_a7j_verbs[] = { @@ -6828,7 +6800,7 @@ static void alc885_macpro_init_hook(struct hda_codec *codec) static void alc885_imac24_init_hook(struct hda_codec *codec) { alc885_macpro_init_hook(codec); - alc885_imac24_automute(codec); + alc885_imac24_automute_init_hook(codec); } /* @@ -6999,8 +6971,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_mbp3_unsol_event, - .init_hook = alc885_mbp3_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc885_mbp3_init_hook, }, [ALC885_MB5] = { .mixers = { alc885_mb5_mixer }, @@ -7036,7 +7008,7 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, - .unsol_event = alc885_imac24_unsol_event, + .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc885_imac24_init_hook, }, [ALC882_TARGA] = { @@ -7053,7 +7025,7 @@ static struct alc_config_preset alc882_presets[] = { .need_dac_fix = 1, .input_mux = &alc882_capture_source, .unsol_event = alc882_targa_unsol_event, - .init_hook = alc882_targa_automute, + .init_hook = alc882_targa_init_hook, }, [ALC882_ASUS_A7J] = { .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, @@ -7903,8 +7875,6 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), @@ -8053,16 +8023,14 @@ static struct hda_verb alc883_init_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc883_mitac_hp_automute(struct hda_codec *codec) +static void alc883_mitac_init_hook(struct hda_codec *codec) { - unsigned int present; + struct alc_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x17; + alc_automute_amp(codec); } /* auto-toggle front mic */ @@ -8079,25 +8047,6 @@ static void alc883_mitac_mic_automute(struct hda_codec *codec) } */ -static void alc883_mitac_automute(struct hda_codec *codec) -{ - alc883_mitac_hp_automute(codec); - /* alc883_mitac_mic_automute(codec); */ -} - -static void alc883_mitac_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_mitac_hp_automute(codec); - break; - case ALC880_MIC_EVENT: - /* alc883_mitac_mic_automute(codec); */ - break; - } -} - static struct hda_verb alc883_mitac_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -8215,29 +8164,15 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; -static void alc888_3st_hp_front_automute(struct hda_codec *codec) +static void alc888_3st_hp_init_hook(struct hda_codec *codec) { - unsigned int present, bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc888_3st_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc888_3st_hp_front_automute(codec); - break; - } + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x18; + alc_automute_amp(codec); } static struct hda_verb alc888_3st_hp_verbs[] = { @@ -8334,56 +8269,18 @@ static struct hda_verb alc883_medion_md2_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc883_medion_md2_automute(struct hda_codec *codec) +static void alc883_medion_md2_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -static void alc883_medion_md2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_medion_md2_automute(codec); -} - -/* toggle speaker-output according to the hp-jack state */ -static void alc883_tagra_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - present ? 1 : 3); -} + struct alc_spec *spec = codec->spec; -static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_tagra_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } /* toggle speaker-output according to the hp-jack state */ -static void alc883_clevo_m720_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} +#define alc883_tagra_init_hook alc882_targa_init_hook +#define alc883_tagra_unsol_event alc882_targa_unsol_event static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { @@ -8395,9 +8292,13 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } -static void alc883_clevo_m720_automute(struct hda_codec *codec) +static void alc883_clevo_m720_init_hook(struct hda_codec *codec) { - alc883_clevo_m720_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); alc883_clevo_m720_mic_automute(codec); } @@ -8405,52 +8306,32 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_clevo_m720_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc883_clevo_m720_mic_automute(codec); break; + default: + alc_automute_amp_unsol_event(codec, res); + break; } } /* toggle speaker-output according to the hp-jack state */ -static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec) +static void alc883_2ch_fujitsu_pi2515_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_2ch_fujitsu_pi2515_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } -static void alc883_haier_w66_automute(struct hda_codec *codec) +static void alc883_haier_w66_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - 0x80, bits); -} + struct alc_spec *spec = codec->spec; -static void alc883_haier_w66_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_haier_w66_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) @@ -8489,23 +8370,14 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, } /* toggle speaker-output according to the hp-jack state */ -static void alc883_acer_aspire_automute(struct hda_codec *codec) +static void alc883_acer_aspire_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc883_acer_aspire_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_acer_aspire_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[1] = 0x16; + alc_automute_amp(codec); } static struct hda_verb alc883_acer_eapd_verbs[] = { @@ -8526,75 +8398,29 @@ static struct hda_verb alc883_acer_eapd_verbs[] = { { } }; -static void alc888_6st_dell_front_automute(struct hda_codec *codec) +static void alc888_6st_dell_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc888_6st_dell_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - /* printk(KERN_DEBUG "hp_event\n"); */ - alc888_6st_dell_front_automute(codec); - break; - } + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.speaker_pins[2] = 0x16; + spec->autocfg.speaker_pins[3] = 0x17; + alc_automute_amp(codec); } -static void alc888_lenovo_sky_front_automute(struct hda_codec *codec) +static void alc888_lenovo_sky_init_hook(struct hda_codec *codec) { - unsigned int mute; - unsigned int present; - - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} + struct alc_spec *spec = codec->spec; -static void alc883_lenovo_sky_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc888_lenovo_sky_front_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.speaker_pins[2] = 0x16; + spec->autocfg.speaker_pins[3] = 0x17; + spec->autocfg.speaker_pins[4] = 0x1a; + alc_automute_amp(codec); } /* @@ -8682,39 +8508,33 @@ static void alc883_nb_mic_automute(struct hda_codec *codec) 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); } -static void alc883_M90V_speaker_automute(struct hda_codec *codec) +static void alc883_M90V_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + struct alc_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); - snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.speaker_pins[2] = 0x16; + alc_automute_pin(codec); } static void alc883_mode2_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_M90V_speaker_automute(codec); - break; case ALC880_MIC_EVENT: alc883_nb_mic_automute(codec); break; + default: + alc_sku_unsol_event(codec, res); + break; } } static void alc883_mode2_inithook(struct hda_codec *codec) { - alc883_M90V_speaker_automute(codec); + alc883_M90V_init_hook(codec); alc883_nb_mic_automute(codec); } @@ -8731,32 +8551,13 @@ static struct hda_verb alc888_asus_eee1601_verbs[] = { { } /* end */ }; -static void alc883_eee1601_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); -} - -static void alc883_eee1601_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_eee1601_speaker_automute(codec); - break; - } -} - static void alc883_eee1601_inithook(struct hda_codec *codec) { - alc883_eee1601_speaker_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc_automute_pin(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -8969,7 +8770,7 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_automute, + .init_hook = alc883_tagra_init_hook, }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_tagra_2ch_mixer}, @@ -8983,7 +8784,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_automute, + .init_hook = alc883_tagra_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, @@ -9008,8 +8809,8 @@ 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_acer_aspire_unsol_event, - .init_hook = alc883_acer_aspire_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_acer_aspire_init_hook, }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_base_mixer, @@ -9028,8 +8829,8 @@ static struct alc_config_preset alc883_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, - .unsol_event = alc888_acer_aspire_4930g_unsol_event, - .init_hook = alc888_acer_aspire_4930g_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_4930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, @@ -9053,8 +8854,8 @@ 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_medion_md2_unsol_event, - .init_hook = alc883_medion_md2_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_medion_md2_init_hook, }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, @@ -9075,7 +8876,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_clevo_m720_unsol_event, - .init_hook = alc883_clevo_m720_automute, + .init_hook = alc883_clevo_m720_init_hook, }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, @@ -9099,8 +8900,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, .input_mux = &alc883_lenovo_nb0763_capture_source, - .unsol_event = alc883_medion_md2_unsol_event, - .init_hook = alc883_medion_md2_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_medion_md2_init_hook, }, [ALC888_LENOVO_MS7195_DIG] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9124,8 +8925,8 @@ 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_haier_w66_unsol_event, - .init_hook = alc883_haier_w66_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_haier_w66_init_hook, }, [ALC888_3ST_HP] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9136,8 +8937,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc888_3st_hp_unsol_event, - .init_hook = alc888_3st_hp_front_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_3st_hp_init_hook, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -9149,8 +8950,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc888_6st_dell_unsol_event, - .init_hook = alc888_6st_dell_front_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_6st_dell_init_hook, }, [ALC883_MITAC] = { .mixers = { alc883_mitac_mixer }, @@ -9160,8 +8961,8 @@ 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_mitac_unsol_event, - .init_hook = alc883_mitac_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_mitac_init_hook, }, [ALC883_FUJITSU_PI2515] = { .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, @@ -9173,8 +8974,8 @@ 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_fujitsu_pi2515_capture_source, - .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, - .init_hook = alc883_2ch_fujitsu_pi2515_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_2ch_fujitsu_pi2515_init_hook, }, [ALC888_FUJITSU_XA3530] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -9191,8 +8992,8 @@ static struct alc_config_preset alc883_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, - .unsol_event = alc888_fujitsu_xa3530_unsol_event, - .init_hook = alc888_fujitsu_xa3530_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_fujitsu_xa3530_init_hook, }, [ALC888_LENOVO_SKY] = { .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, @@ -9204,8 +9005,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .need_dac_fix = 1, .input_mux = &alc883_lenovo_sky_capture_source, - .unsol_event = alc883_lenovo_sky_unsol_event, - .init_hook = alc888_lenovo_sky_front_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_lenovo_sky_init_hook, }, [ALC888_ASUS_M90V] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9233,7 +9034,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, .input_mux = &alc883_asus_eee1601_capture_source, - .unsol_event = alc883_eee1601_unsol_event, + .unsol_event = alc_sku_unsol_event, .init_hook = alc883_eee1601_inithook, }, [ALC1200_ASUS_P5Q] = { @@ -9666,28 +9467,15 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hp_t5735_automute(struct hda_codec *codec) +static void alc262_hp_t5735_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE, - spec->jack_present ? HDA_AMP_MUTE : 0); -} - -static void alc262_hp_t5735_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hp_t5735_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + alc_automute_amp(codec); } -#define alc262_hp_t5735_init_hook alc262_hp_t5735_automute - static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9914,34 +9702,15 @@ static struct hda_verb alc262_tyan_verbs[] = { }; /* unsolicited event for HP jack sensing */ -static void alc262_tyan_automute(struct hda_codec *codec) +static void alc262_tyan_init_hook(struct hda_codec *codec) { - unsigned int mute; - unsigned int present; + struct alc_spec *spec = codec->spec; - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute line output on ATX panel */ - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute line output if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } -static void alc262_tyan_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_tyan_automute(codec); -} #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -10096,35 +9865,24 @@ static void alc262_dmic_automute(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09); } -/* toggle speaker-output according to the hp-jack state */ -static void alc262_toshiba_s06_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, bits); -} - - /* unsolicited event for HP jack sensing */ static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc262_toshiba_s06_speaker_automute(codec); if ((res >> 26) == ALC880_MIC_EVENT) alc262_dmic_automute(codec); - + else + alc_sku_unsol_event(codec, res); } static void alc262_toshiba_s06_init_hook(struct hda_codec *codec) { - alc262_toshiba_s06_speaker_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_pin(codec); alc262_dmic_automute(codec); } @@ -11135,7 +10893,7 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hp_t5735_unsol_event, + .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc262_hp_t5735_init_hook, }, [ALC262_HP_RP5700] = { @@ -11256,8 +11014,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_tyan_unsol_event, - .init_hook = alc262_tyan_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc262_tyan_init_hook, }, }; @@ -11646,30 +11404,15 @@ static struct hda_verb alc268_dell_verbs[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc268_dell_automute(struct hda_codec *codec) +static void alc268_dell_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned int mute; - - present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0); - if (present & 0x80000000) - mute = HDA_AMP_MUTE; - else - mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); -} + struct alc_spec *spec = codec->spec; -static void alc268_dell_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc268_dell_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_pin(codec); } -#define alc268_dell_init_hook alc268_dell_automute - static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -11688,16 +11431,6 @@ static struct hda_verb alc267_quanta_il1_verbs[] = { { } }; -static void alc267_quanta_il1_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - present ? 0 : PIN_OUT); -} - static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -11709,9 +11442,13 @@ static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) present ? 0x00 : 0x01); } -static void alc267_quanta_il1_automute(struct hda_codec *codec) +static void alc267_quanta_il1_init_hook(struct hda_codec *codec) { - alc267_quanta_il1_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_pin(codec); alc267_quanta_il1_mic_automute(codec); } @@ -11719,12 +11456,12 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc267_quanta_il1_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc267_quanta_il1_mic_automute(codec); break; + default: + alc_sku_unsol_event(codec, res); + break; } } @@ -12198,7 +11935,7 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc267_quanta_il1_unsol_event, - .init_hook = alc267_quanta_il1_automute, + .init_hook = alc267_quanta_il1_init_hook, }, [ALC268_3ST] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, @@ -12293,7 +12030,7 @@ static struct alc_config_preset alc268_presets[] = { .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .unsol_event = alc268_dell_unsol_event, + .unsol_event = alc_sku_unsol_event, .init_hook = alc268_dell_init_hook, .input_mux = &alc268_capture_source, }, @@ -14727,19 +14464,6 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {} }; -/* toggle speaker-output according to the hp-jack state */ -static void alc861vd_lenovo_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -14752,9 +14476,13 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); } -static void alc861vd_lenovo_automute(struct hda_codec *codec) +static void alc861vd_lenovo_init_hook(struct hda_codec *codec) { - alc861vd_lenovo_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); alc861vd_lenovo_mic_automute(codec); } @@ -14762,12 +14490,12 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc861vd_lenovo_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc861vd_lenovo_mic_automute(codec); break; + default: + alc_automute_amp_unsol_event(codec, res); + break; } } @@ -14817,20 +14545,13 @@ static struct hda_verb alc861vd_dallas_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc861vd_dallas_automute(struct hda_codec *codec) +static void alc861vd_dallas_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc861vd_dallas_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -14944,7 +14665,7 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, .unsol_event = alc861vd_lenovo_unsol_event, - .init_hook = alc861vd_lenovo_automute, + .init_hook = alc861vd_lenovo_init_hook, }, [ALC861VD_DALLAS] = { .mixers = { alc861vd_dallas_mixer }, @@ -14954,8 +14675,8 @@ static struct alc_config_preset alc861vd_presets[] = { .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_dallas_capture_source, - .unsol_event = alc861vd_dallas_unsol_event, - .init_hook = alc861vd_dallas_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc861vd_dallas_init_hook, }, [ALC861VD_HP] = { .mixers = { alc861vd_hp_mixer }, @@ -14966,8 +14687,8 @@ static struct alc_config_preset alc861vd_presets[] = { .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_hp_capture_source, - .unsol_event = alc861vd_dallas_unsol_event, - .init_hook = alc861vd_dallas_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc861vd_dallas_init_hook, }, [ALC660VD_ASUS_V1S] = { .mixers = { alc861vd_lenovo_mixer }, @@ -14982,7 +14703,7 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, .unsol_event = alc861vd_lenovo_unsol_event, - .init_hook = alc861vd_lenovo_automute, + .init_hook = alc861vd_lenovo_init_hook, }, }; -- cgit v1.2.3 From 01f2bd48d08c6bbde12d86b66c760612e33e49a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 May 2009 08:12:43 +0200 Subject: ALSA: hda - Add missing models for Realtek codecs Added the missing descriptions and the model names for Realtek codecs to the documentation and the config table. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 10 ++++++++-- sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 36c97126d17d..92c9d3a3e990 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -36,6 +36,7 @@ ALC260 acer Acer TravelMate will Will laptops (PB V7900) replacer Replacer 672V + favorit100 Maxdata Favorit 100XS basic fixed pin assignment (old default model) test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with @@ -85,10 +86,11 @@ ALC269 eeepc-p703 ASUS Eeepc P703 P900A eeepc-p901 ASUS Eeepc P901 S101 fujitsu FSC Amilo + lifebook Fujitsu Lifebook S6420 auto auto-config reading BIOS (default) -ALC662/663 -========== +ALC662/663/272 +============== 3stack-dig 3-stack (2-channel) with SPDIF 3stack-6ch 3-stack (6-channel) 3stack-6ch-dig 3-stack (6-channel) with SPDIF @@ -107,6 +109,8 @@ ALC662/663 asus-mode4 ASUS asus-mode5 ASUS asus-mode6 ASUS + dell Dell with ALC272 + dell-zm1 Dell ZM1 with ALC272 auto auto-config reading BIOS (default) ALC882/885 @@ -118,6 +122,7 @@ ALC882/885 asus-a7j ASUS A7J asus-a7m ASUS A7M macpro MacPro support + mb5 Macbook 5,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC @@ -150,6 +155,7 @@ ALC883/888 fujitsu-pi2515 Fujitsu AMILO Pi2515 fujitsu-xa3530 Fujitsu AMILO XA3530 3stack-6ch-intel Intel DG33* boards + asus-p5q ASUS P5Q-EM boards auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34f6fb72f2bc..db896121535a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16184,6 +16184,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC663_ASUS_MODE4] = "asus-mode4", [ALC663_ASUS_MODE5] = "asus-mode5", [ALC663_ASUS_MODE6] = "asus-mode6", + [ALC272_DELL] = "dell", + [ALC272_DELL_ZM1] = "dell-zm1", [ALC662_AUTO] = "auto", }; -- cgit v1.2.3 From 9541ba1d665542c96f7c0b5b836bbc1fd9d961b6 Mon Sep 17 00:00:00 2001 From: Chris Pockelé Date: Tue, 12 May 2009 08:08:53 +0200 Subject: ALSA: hda - Add support of Samsung NC10 mini notebook MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add specific configuration for Samsung NC10 mini notebook. Internal mic/speakers will be correctly muted when plugging in external ones. Mixer controls are added for speakers, headphones and PC beep. "Boost" is added for the microphones. Signed-off-by: Chris Pockelé Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 64 ++++++++++++++++++++++++++++ 2 files changed, 65 insertions(+) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 92c9d3a3e990..14ed1d114d7a 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -111,6 +111,7 @@ ALC662/663/272 asus-mode6 ASUS dell Dell with ALC272 dell-zm1 Dell ZM1 with ALC272 + samsung-nc10 Samsung NC10 mini notebook auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index db896121535a..c14020ab3cb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -190,6 +190,7 @@ enum { ALC663_ASUS_MODE6, ALC272_DELL, ALC272_DELL_ZM1, + ALC272_SAMSUNG_NC10, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -15120,6 +15121,38 @@ static struct hda_input_mux alc663_m51va_capture_source = { }, }; +#if 1 /* set to 0 for testing other input sources below */ +static struct hda_input_mux alc272_nc10_capture_source = { + .num_items = 2, + .items = { + { "Autoselect Mic", 0x0 }, + { "Internal Mic", 0x1 }, + }, +}; +#else +static struct hda_input_mux alc272_nc10_capture_source = { + .num_items = 16, + .items = { + { "Autoselect Mic", 0x0 }, + { "Internal Mic", 0x1 }, + { "In-0x02", 0x2 }, + { "In-0x03", 0x3 }, + { "In-0x04", 0x4 }, + { "In-0x05", 0x5 }, + { "In-0x06", 0x6 }, + { "In-0x07", 0x7 }, + { "In-0x08", 0x8 }, + { "In-0x09", 0x9 }, + { "In-0x0a", 0x0a }, + { "In-0x0b", 0x0b }, + { "In-0x0c", 0x0c }, + { "In-0x0d", 0x0d }, + { "In-0x0e", 0x0e }, + { "In-0x0f", 0x0f }, + }, +}; +#endif + /* * 2ch mode */ @@ -16151,6 +16184,23 @@ static struct snd_kcontrol_new alc662_ecs_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc272_nc10_mixer[] = { + /* Master Playback automatically created from Speaker and Headphone */ + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -16186,6 +16236,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC663_ASUS_MODE6] = "asus-mode6", [ALC272_DELL] = "dell", [ALC272_DELL_ZM1] = "dell-zm1", + [ALC272_SAMSUNG_NC10] = "samsung-nc10", [ALC662_AUTO] = "auto", }; @@ -16243,6 +16294,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), @@ -16514,6 +16566,18 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc663_m51va_unsol_event, .init_hook = alc663_m51va_inithook, }, + [ALC272_SAMSUNG_NC10] = { + .mixers = { alc272_nc10_mixer }, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc272_dac_nids), + .dac_nids = alc272_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc272_nc10_capture_source, + .unsol_event = alc663_mode4_unsol_event, + .init_hook = alc663_mode4_inithook, + }, }; -- cgit v1.2.3 From 22b530e0287724f00dde9a15a94d408c3334e86c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 May 2009 11:32:52 +0200 Subject: ALSA: hda - Disable fallback to model=acer for Acer laptops The model=acer for ALC883/889 doesn't work well for the recent Acer Aspire laptops. Since model=auto works better nowadays, it's safer to use the default fallback instead of the Acer specific one. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c14020ab3cb6..51954bacaa01 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8622,8 +8622,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_4930G), - /* default Acer */ - SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), + /* default Acer -- disabled as it causes more problems. + * model=auto should work fine now + */ + /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), -- cgit v1.2.3 From 7442f9dadb8626bd9e7e01445560499560b99f5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 18:20:33 +0200 Subject: ALSA: hda - Add a quirk entry for Macbook Pro 5,1 Added the codec SSID for MacBook Pro 5,1 as compatible as MP51. However, the headphone auto-muting function doesn't work. So, this is just a tentative solution. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 51954bacaa01..1bf9d054aae6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7293,6 +7293,10 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MBP3; break; case 0x106b3f00: /* Macbook 5,1 */ + case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense + * seems not working, so apparently + * no perfect solution yet + */ board_config = ALC885_MB5; break; default: -- cgit v1.2.3 From 2fc998907947f92b1b9113faad531d7f5a857987 Mon Sep 17 00:00:00 2001 From: Nickolas Lloyd Date: Fri, 15 May 2009 15:33:30 +0200 Subject: ALSA: hda - add controls to toggle DC bias on mic ports This patch adds a mixer control for the STAC92XX boards to control the DC bias of mic ports, allowing recording from both powered and non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack Mode" to switch between Mic, Line In, and Line Out. Signed-off-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 111 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 107 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ecf53f755a09..02950980778e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -634,6 +634,96 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } +static unsigned int stac92xx_vref_set(struct hda_codec *codec, + hda_nid_t nid, unsigned int new_vref) +{ + unsigned int error; + unsigned int pincfg; + pincfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + pincfg &= 0xff; + pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + pincfg |= new_vref; + + if (new_vref == AC_PINCTL_VREF_HIZ) + pincfg |= AC_PINCTL_OUT_EN; + else + pincfg |= AC_PINCTL_IN_EN; + + error = snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg); + if (error < 0) + return error; + else + return 1; +} + +static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int vref; + vref = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + vref &= AC_PINCTL_VREFEN; + return vref; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref; + unsigned int error; + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = AC_PINCTL_VREF_80; + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else + new_vref = AC_PINCTL_VREF_HIZ; + + if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { + error = stac92xx_vref_set(codec, + kcontrol->private_value, new_vref); + return error; + } + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); + if (vref == AC_PINCTL_VREF_80) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 3; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -995,6 +1085,17 @@ static struct hda_verb stac9205_core_init[] = { .private_value = verb_read | (verb_write << 16), \ } +#define DC_BIAS(xname, idx, nid) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = idx, \ + .info = stac92xx_dc_bias_info, \ + .get = stac92xx_dc_bias_get, \ + .put = stac92xx_dc_bias_put, \ + .private_value = nid, \ + } + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -2702,7 +2803,8 @@ enum { STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, - STAC_CTL_WIDGET_CLFE_SWITCH + STAC_CTL_WIDGET_CLFE_SWITCH, + STAC_CTL_WIDGET_DC_BIAS }; static struct snd_kcontrol_new stac92xx_control_templates[] = { @@ -2714,6 +2816,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), + DC_BIAS(NULL, 0, 0), }; /* add dynamic controls */ @@ -3165,9 +3268,9 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", - (spec->mic_switch << 8) | 1); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, + "Mic Jack Mode", + spec->mic_switch); if (err < 0) return err; } -- cgit v1.2.3 From 812a2cca295ee7f56cd1b988a0f93646285c214a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 16 May 2009 10:00:49 +0200 Subject: ALSA: hda - Split codec->name to vendor and chip name strings Split the name string in hda_codec struct to vendor_name and chip_name strings to be stored directly from the preset name. Since mostly only the chip name is referred in many patch_*.c, this results in the reduction of many codes in the end. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 36 ++++++++++++++++--------- sound/pci/hda/hda_codec.h | 3 ++- sound/pci/hda/hda_hwdep.c | 9 ++++--- sound/pci/hda/hda_proc.c | 8 ++++-- sound/pci/hda/patch_realtek.c | 63 ++++++------------------------------------- 5 files changed, 46 insertions(+), 73 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b91f6ed5cc58..77385de01744 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -623,7 +623,10 @@ static int get_codec_name(struct hda_codec *codec) const struct hda_vendor_id *c; const char *vendor = NULL; u16 vendor_id = codec->vendor_id >> 16; - char tmp[16], name[32]; + char tmp[16]; + + if (codec->vendor_name) + goto get_chip_name; for (c = hda_vendor_ids; c->id; c++) { if (c->id == vendor_id) { @@ -635,14 +638,21 @@ static int get_codec_name(struct hda_codec *codec) sprintf(tmp, "Generic %04x", vendor_id); vendor = tmp; } + codec->vendor_name = kstrdup(vendor, GFP_KERNEL); + if (!codec->vendor_name) + return -ENOMEM; + + get_chip_name: + if (codec->chip_name) + return 0; + if (codec->preset && codec->preset->name) - snprintf(name, sizeof(name), "%s %s", vendor, - codec->preset->name); - else - snprintf(name, sizeof(name), "%s ID %x", vendor, - codec->vendor_id & 0xffff); - codec->name = kstrdup(name, GFP_KERNEL); - if (!codec->name) + codec->chip_name = kstrdup(codec->preset->name, GFP_KERNEL); + else { + sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); + codec->chip_name = kstrdup(tmp, GFP_KERNEL); + } + if (!codec->chip_name) return -ENOMEM; return 0; } @@ -848,7 +858,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); - kfree(codec->name); + kfree(codec->vendor_name); + kfree(codec->chip_name); kfree(codec->modelname); kfree(codec->wcaps); kfree(codec); @@ -989,15 +1000,16 @@ int snd_hda_codec_configure(struct hda_codec *codec) int err; codec->preset = find_codec_preset(codec); - if (!codec->name) { + if (!codec->vendor_name || !codec->chip_name) { err = get_codec_name(codec); if (err < 0) return err; } /* audio codec should override the mixer name */ if (codec->afg || !*codec->bus->card->mixername) - strlcpy(codec->bus->card->mixername, codec->name, - sizeof(codec->bus->card->mixername)); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); if (is_generic_config(codec)) { err = snd_hda_parse_generic_codec(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index cd8979c7670b..c5bd40f77bb9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -748,7 +748,8 @@ struct hda_codec { /* detected preset */ const struct hda_codec_preset *preset; struct module *owner; - const char *name; /* codec name */ + const char *vendor_name; /* codec vendor name */ + const char *chip_name; /* codec chip name */ const char *modelname; /* model name for preset */ /* set by patch */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 1c57505c2874..6812fbe80fa4 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -242,7 +242,8 @@ CODEC_INFO_SHOW(subsystem_id); CODEC_INFO_SHOW(revision_id); CODEC_INFO_SHOW(afg); CODEC_INFO_SHOW(mfg); -CODEC_INFO_STR_SHOW(name); +CODEC_INFO_STR_SHOW(vendor_name); +CODEC_INFO_STR_SHOW(chip_name); CODEC_INFO_STR_SHOW(modelname); #define CODEC_INFO_STORE(type) \ @@ -275,7 +276,8 @@ static ssize_t type##_store(struct device *dev, \ CODEC_INFO_STORE(vendor_id); CODEC_INFO_STORE(subsystem_id); CODEC_INFO_STORE(revision_id); -CODEC_INFO_STR_STORE(name); +CODEC_INFO_STR_STORE(vendor_name); +CODEC_INFO_STR_STORE(chip_name); CODEC_INFO_STR_STORE(modelname); #define CODEC_ACTION_STORE(type) \ @@ -499,7 +501,8 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RW(revision_id), CODEC_ATTR_RO(afg), CODEC_ATTR_RO(mfg), - CODEC_ATTR_RW(name), + CODEC_ATTR_RW(vendor_name), + CODEC_ATTR_RW(chip_name), CODEC_ATTR_RW(modelname), CODEC_ATTR_RW(init_verbs), CODEC_ATTR_RW(hints), diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 93d7499350c6..418c5d1badaa 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -466,8 +466,12 @@ static void print_codec_info(struct snd_info_entry *entry, hda_nid_t nid; int i, nodes; - snd_iprintf(buffer, "Codec: %s\n", - codec->name ? codec->name : "Not Set"); + snd_iprintf(buffer, "Codec: "); + if (codec->vendor_name && codec->chip_name) + snd_iprintf(buffer, "%s %s\n", + codec->vendor_name, codec->chip_name); + else + snd_iprintf(buffer, "Not Set\n"); snd_iprintf(buffer, "Address: %d\n", codec->addr); snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1bf9d054aae6..4bc3e7e53555 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -277,13 +277,13 @@ struct alc_spec { */ unsigned int num_init_verbs; - char *stream_name_analog; /* analog PCM stream */ + char stream_name_analog[16]; /* analog PCM stream */ struct hda_pcm_stream *stream_analog_playback; struct hda_pcm_stream *stream_analog_capture; struct hda_pcm_stream *stream_analog_alt_playback; struct hda_pcm_stream *stream_analog_alt_capture; - char *stream_name_digital; /* digital PCM stream */ + char stream_name_digital[16]; /* digital PCM stream */ struct hda_pcm_stream *stream_digital_playback; struct hda_pcm_stream *stream_digital_capture; @@ -3169,7 +3169,10 @@ static int alc_build_pcms(struct hda_codec *codec) if (spec->no_analog) goto skip_analog; + snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog), + "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; + if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) return -EINVAL; @@ -3195,6 +3198,9 @@ static int alc_build_pcms(struct hda_codec *codec) skip_analog: /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { + snprintf(spec->stream_name_digital, + sizeof(spec->stream_name_digital), + "%s Digital", codec->chip_name); codec->num_pcms = 2; codec->slave_dig_outs = spec->multiout.slave_dig_outs; info = spec->pcm_rec + 1; @@ -4432,12 +4438,10 @@ static int patch_alc880(struct hda_codec *codec) if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); - spec->stream_name_analog = "ALC880 Analog"; spec->stream_analog_playback = &alc880_pcm_analog_playback; spec->stream_analog_capture = &alc880_pcm_analog_capture; spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC880 Digital"; spec->stream_digital_playback = &alc880_pcm_digital_playback; spec->stream_digital_capture = &alc880_pcm_digital_capture; @@ -6078,11 +6082,9 @@ static int patch_alc260(struct hda_codec *codec) if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); - spec->stream_name_analog = "ALC260 Analog"; spec->stream_analog_playback = &alc260_pcm_analog_playback; spec->stream_analog_capture = &alc260_pcm_analog_capture; - spec->stream_name_digital = "ALC260 Digital"; spec->stream_digital_playback = &alc260_pcm_digital_playback; spec->stream_digital_capture = &alc260_pcm_digital_capture; @@ -7337,14 +7339,6 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (codec->vendor_id == 0x10ec0885) { - spec->stream_name_analog = "ALC885 Analog"; - spec->stream_name_digital = "ALC885 Digital"; - } else { - spec->stream_name_analog = "ALC882 Analog"; - spec->stream_name_digital = "ALC882 Digital"; - } - spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; /* FIXME: setup DAC5 */ @@ -9232,13 +9226,6 @@ static int patch_alc883(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0888: - if (codec->revision_id == 0x100101) { - spec->stream_name_analog = "ALC1200 Analog"; - spec->stream_name_digital = "ALC1200 Digital"; - } else { - spec->stream_name_analog = "ALC888 Analog"; - spec->stream_name_digital = "ALC888 Digital"; - } if (!spec->num_adc_nids) { spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); spec->adc_nids = alc883_adc_nids; @@ -9249,8 +9236,6 @@ static int patch_alc883(struct hda_codec *codec) spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ break; case 0x10ec0889: - spec->stream_name_analog = "ALC889 Analog"; - spec->stream_name_digital = "ALC889 Digital"; if (!spec->num_adc_nids) { spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); spec->adc_nids = alc889_adc_nids; @@ -9261,8 +9246,6 @@ static int patch_alc883(struct hda_codec *codec) capture */ break; default: - spec->stream_name_analog = "ALC883 Analog"; - spec->stream_name_digital = "ALC883 Digital"; if (!spec->num_adc_nids) { spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); spec->adc_nids = alc883_adc_nids; @@ -11087,11 +11070,9 @@ static int patch_alc262(struct hda_codec *codec) if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); - spec->stream_name_analog = "ALC262 Analog"; spec->stream_analog_playback = &alc262_pcm_analog_playback; spec->stream_analog_capture = &alc262_pcm_analog_capture; - spec->stream_name_digital = "ALC262 Digital"; spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; @@ -12117,14 +12098,6 @@ static int patch_alc268(struct hda_codec *codec) if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); - if (codec->vendor_id == 0x10ec0267) { - spec->stream_name_analog = "ALC267 Analog"; - spec->stream_name_digital = "ALC267 Digital"; - } else { - spec->stream_name_analog = "ALC268 Analog"; - spec->stream_name_digital = "ALC268 Digital"; - } - spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture; @@ -12979,7 +12952,6 @@ static int patch_alc269(struct hda_codec *codec) if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); - spec->stream_name_analog = "ALC269 Analog"; if (codec->subsystem_id == 0x17aa3bf8) { /* Due to a hardware problem on Lenovo Ideadpad, we need to * fix the sample rate of analog I/O to 44.1kHz @@ -12990,7 +12962,6 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_analog_playback = &alc269_pcm_analog_playback; spec->stream_analog_capture = &alc269_pcm_analog_capture; } - spec->stream_name_digital = "ALC269 Digital"; spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; @@ -14080,11 +14051,9 @@ static int patch_alc861(struct hda_codec *codec) if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); - spec->stream_name_analog = "ALC861 Analog"; spec->stream_analog_playback = &alc861_pcm_analog_playback; spec->stream_analog_capture = &alc861_pcm_analog_capture; - spec->stream_name_digital = "ALC861 Digital"; spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; @@ -15007,13 +14976,8 @@ static int patch_alc861vd(struct hda_codec *codec) setup_preset(spec, &alc861vd_presets[board_config]); if (codec->vendor_id == 0x10ec0660) { - spec->stream_name_analog = "ALC660-VD Analog"; - spec->stream_name_digital = "ALC660-VD Digital"; /* always turn on EAPD */ add_verb(spec, alc660vd_eapd_verbs); - } else { - spec->stream_name_analog = "ALC861VD Analog"; - spec->stream_name_digital = "ALC861VD Digital"; } spec->stream_analog_playback = &alc861vd_pcm_analog_playback; @@ -16936,17 +16900,6 @@ static int patch_alc662(struct hda_codec *codec) if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); - if (codec->vendor_id == 0x10ec0663) { - spec->stream_name_analog = "ALC663 Analog"; - spec->stream_name_digital = "ALC663 Digital"; - } else if (codec->vendor_id == 0x10ec0272) { - spec->stream_name_analog = "ALC272 Analog"; - spec->stream_name_digital = "ALC272 Digital"; - } else { - spec->stream_name_analog = "ALC662 Analog"; - spec->stream_name_digital = "ALC662 Digital"; - } - spec->stream_analog_playback = &alc662_pcm_analog_playback; spec->stream_analog_capture = &alc662_pcm_analog_capture; -- cgit v1.2.3 From 6c627f3978bf3059d45713e2e1240b7c43cc85f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 May 2009 12:33:36 +0200 Subject: ALSA: hda - Show the actual chip name in 'unkown model' messages Show the actual chip name in 'unknown model..' info messages for Realtek codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4bc3e7e53555..62f4d0ca7e14 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4410,8 +4410,8 @@ static int patch_alc880(struct hda_codec *codec) alc880_models, alc880_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC880, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC880_AUTO; } @@ -6054,8 +6054,9 @@ static int patch_alc260(struct hda_codec *codec) alc260_models, alc260_cfg_tbl); if (board_config < 0) { - snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, " - "trying auto-probe from BIOS...\n"); + snd_printd(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", + codec->chip_name); board_config = ALC260_AUTO; } @@ -7308,8 +7309,9 @@ static int patch_alc882(struct hda_codec *codec) alc_free(codec); return patch_alc883(codec); } - printk(KERN_INFO "hda_codec: Unknown model for ALC882, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", + codec->chip_name); board_config = ALC882_AUTO; } } @@ -9196,8 +9198,8 @@ static int patch_alc883(struct hda_codec *codec) alc883_models, alc883_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC883, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC883_AUTO; } @@ -11040,8 +11042,8 @@ static int patch_alc262(struct hda_codec *codec) alc262_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC262, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC262_AUTO; } @@ -12076,8 +12078,8 @@ static int patch_alc268(struct hda_codec *codec) alc268_cfg_tbl); if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for ALC268, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC268_AUTO; } @@ -12924,8 +12926,8 @@ static int patch_alc269(struct hda_codec *codec) alc269_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC269, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC269_AUTO; } @@ -14023,8 +14025,8 @@ static int patch_alc861(struct hda_codec *codec) alc861_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC861, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC861_AUTO; } @@ -14947,8 +14949,8 @@ static int patch_alc861vd(struct hda_codec *codec) alc861vd_cfg_tbl); if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for ALC660VD/" - "ALC861VD, trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC861VD_AUTO; } @@ -16872,8 +16874,8 @@ static int patch_alc662(struct hda_codec *codec) alc662_models, alc662_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC662, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC662_AUTO; } -- cgit v1.2.3 From 313f6e2d40bd67e394a65e7d64acd0cf9c9d248d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 May 2009 12:40:52 +0200 Subject: ALSA: hda - Avoid conflicts with snd-ctxfi driver The PCI entries of Creative with HD-audio class can be the devices with emu20k1/emu20k2 chips. These are supported better by snd-ctxfi driver. With that driver, the device will mutate from HD-audio to its native class. This patch adds a simple ifdef to avoid the conflict of device probe between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems still OK to be added as it has no emu20kx chip, and is a pure HD-audio device. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 04f19f8cad84..7bb6dd2cb6cc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2517,10 +2517,19 @@ static struct pci_device_id azx_ids[] = { /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* Creative X-Fi (CA0110-IBG) */ +#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) + /* the following entry conflicts with snd-ctxfi driver, + * as ctxfi driver mutates from HD-audio to native mode with + * a special command sequence. + */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_GENERIC }, +#else + /* this entry seems still valid -- i.e. without emu20kx chip */ + { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, +#endif /* AMD Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- cgit v1.2.3 From eb4c41d30ba1f973c8b10be3644571ab997ae0d6 Mon Sep 17 00:00:00 2001 From: Torben Schulz Date: Mon, 18 May 2009 15:02:35 +0200 Subject: ALSA: hda - Improved MacBook 3,1 support This patch adds support for MacBook 3,1 sound by adding a model new "mb31" with the appropriate init verbs, mixers and channel modes to the ALC883 configuration. patch_alc882() and patch_alc883() are modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A) correctly. Signed-off-by: Torben Schulz Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 151 ++++++++++++++++++++++++++- 2 files changed, 147 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 14ed1d114d7a..ebb444d5282f 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -157,6 +157,7 @@ ALC883/888 fujitsu-xa3530 Fujitsu AMILO XA3530 3stack-6ch-intel Intel DG33* boards asus-p5q ASUS P5Q-EM boards + mb31 MacBook 3,1 auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62f4d0ca7e14..b942019461a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -240,6 +240,7 @@ enum { ALC883_3ST_6ch_INTEL, ALC888_ASUS_M90V, ALC888_ASUS_EEE1601, + ALC889A_MB31, ALC1200_ASUS_P5Q, ALC883_AUTO, ALC883_MODEL_LAST, @@ -7291,7 +7292,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ - case 0x106b3600: /* Macbook 3.1 */ + /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */ case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; @@ -7493,6 +7494,17 @@ static struct hda_input_mux alc883_asus_eee1601_capture_source = { }, }; +static struct hda_input_mux alc889A_mb31_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + /* Front Mic (0x01) unused */ + { "Line", 0x2 }, + /* Line 2 (0x03) unused */ + /* CD (0x04) unsused? */ + }, +}; + /* * 2ch mode */ @@ -7611,6 +7623,49 @@ static struct hda_channel_mode alc883_sixstack_modes[2] = { { 8, alc883_sixstack_ch8_init }, }; +/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ +static struct hda_verb alc889A_mb31_ch2_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ + { } /* end */ +}; + +/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ +static struct hda_verb alc889A_mb31_ch4_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ + { } /* end */ +}; + +/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ +static struct hda_verb alc889A_mb31_ch5_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ + { } /* end */ +}; + +/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ +static struct hda_verb alc889A_mb31_ch6_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ + { } /* end */ +}; + +static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { + { 2, alc889A_mb31_ch2_init }, + { 4, alc889A_mb31_ch4_init }, + { 5, alc889A_mb31_ch5_init }, + { 6, alc889A_mb31_ch6_init }, +}; + static struct hda_verb alc883_medion_eapd_verbs[] = { /* eanable EAPD on medion laptop */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -7889,6 +7944,32 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc889A_mb31_mixer[] = { + /* Output mixers */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT), + /* Output switches */ + HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), + /* Boost mixers */ + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT), + /* Input mixers */ + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + static struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -8561,6 +8642,42 @@ static void alc883_eee1601_inithook(struct hda_codec *codec) alc_automute_pin(codec); } +static struct hda_verb alc889A_mb31_verbs[] = { + /* Init rear pin (used as headphone output) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Init line pin (used as output in 4ch and 6ch mode) */ + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */ + /* Init line 2 pin (used as headphone out by default) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */ + { } /* end */ +}; + +/* Mute speakers according to the headphone jack state */ +static void alc889A_mb31_automute(struct hda_codec *codec) +{ + unsigned int present; + + /* Mute only in 2ch or 4ch mode */ + if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) + == 0x00) { + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + } +} + +static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc889A_mb31_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc883_loopbacks alc880_loopbacks #endif @@ -8601,6 +8718,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC1200_ASUS_P5Q] = "asus-p5q", + [ALC889A_MB31] = "mb31", [ALC883_AUTO] = "auto", }; @@ -9052,6 +9170,21 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC889A_MB31] = { + .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, + .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, + alc880_gpio1_init_verbs }, + .adc_nids = alc883_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .dac_nids = alc883_dac_nids, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .channel_mode = alc889A_mb31_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes), + .input_mux = &alc889A_mb31_capture_source, + .dig_out_nid = ALC883_DIGOUT_NID, + .unsol_event = alc889A_mb31_unsol_event, + .init_hook = alc889A_mb31_automute, + }, }; @@ -9197,10 +9330,18 @@ static int patch_alc883(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, alc883_models, alc883_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); - board_config = ALC883_AUTO; + if (board_config < 0 || board_config >= ALC883_MODEL_LAST) { + /* Pick up systems that don't supply PCI SSID */ + switch (codec->subsystem_id) { + case 0x106b3600: /* Macbook 3.1 */ + board_config = ALC889A_MB31; + break; + default: + printk(KERN_INFO + "hda_codec: Unknown model for %s, trying " + "auto-probe from BIOS...\n", codec->chip_name); + board_config = ALC883_AUTO; + } } if (board_config == ALC883_AUTO) { -- cgit v1.2.3 From 4abc1cc2f9fe4b6bb3acc1d78e2c15af47b8133d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 May 2009 12:16:46 +0200 Subject: ALSA: hda - Add prefix to kernel messages Add proper prefix to each kernel message in hda_intel.c. Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used together with snd_print*(). Reference: bko#13207 http://bugzilla.kernel.org/show_bug.cgi?id=13207 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7bb6dd2cb6cc..49fd973b85cc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -128,8 +128,11 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{ULI, M5461}}"); MODULE_DESCRIPTION("Intel HDA driver"); +#ifdef CONFIG_SND_VERBOSE_PRINTK +#define SFX /* nop */ +#else #define SFX "hda-intel: " - +#endif /* * registers @@ -620,7 +623,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) } if (chip->msi) { - snd_printk(KERN_WARNING "hda_intel: No response from codec, " + snd_printk(KERN_WARNING SFX "No response from codec, " "disabling MSI: last cmd=0x%08x\n", chip->last_cmd); free_irq(chip->irq, chip); chip->irq = -1; @@ -634,7 +637,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) } if (!chip->polling_mode) { - snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, " + snd_printk(KERN_WARNING SFX "azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", chip->last_cmd); chip->polling_mode = 1; @@ -649,7 +652,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) return -1; } - snd_printk(KERN_ERR "hda_intel: azx_get_response timeout (ERROR): " + snd_printk(KERN_ERR SFX "azx_get_response timeout (ERROR): " "last cmd=0x%08x\n", chip->last_cmd); spin_lock_irq(&chip->reg_lock); chip->rirb.cmds = 0; /* reset the index */ @@ -776,7 +779,7 @@ static int azx_reset(struct azx *chip) /* check to see if controller is ready */ if (!azx_readb(chip, GCTL)) { - snd_printd("azx_reset: controller not ready!\n"); + snd_printd(SFX "azx_reset: controller not ready!\n"); return -EBUSY; } @@ -786,7 +789,7 @@ static int azx_reset(struct azx *chip) /* detect codecs */ if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); - snd_printdd("codec_mask = 0x%x\n", chip->codec_mask); + snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask); } return 0; @@ -954,12 +957,12 @@ static void azx_init_pci(struct azx *chip) case AZX_DRIVER_SCH: pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) { - pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, \ + pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP)); pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); - snd_printdd("HDA snoop disabled, enabling ... %s\n",\ - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) \ + snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n", + (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) ? "Failed" : "OK"); } break; @@ -1099,7 +1102,7 @@ static int azx_setup_periods(struct azx *chip, pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { - snd_printk(KERN_WARNING "Too big adjustment %d\n", + snd_printk(KERN_WARNING SFX "Too big adjustment %d\n", bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { @@ -1123,7 +1126,7 @@ static int azx_setup_periods(struct azx *chip, return 0; error: - snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", + snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); return -EINVAL; } @@ -1216,7 +1219,7 @@ static int probe_codec(struct azx *chip, int addr) chip->probing = 0; if (res == -1) return -EIO; - snd_printdd("hda_intel: codec #%d probed OK\n", addr); + snd_printdd(SFX "codec #%d probed OK\n", addr); return 0; } @@ -1271,8 +1274,8 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* Some BIOSen give you wrong codec addresses * that don't exist */ - snd_printk(KERN_WARNING - "hda_intel: Codec #%d probe error; " + snd_printk(KERN_WARNING SFX + "Codec #%d probe error; " "disabling it...\n", c); chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing @@ -1488,7 +1491,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) bufsize = snd_pcm_lib_buffer_bytes(substream); period_bytes = snd_pcm_lib_period_bytes(substream); - snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", bufsize, format_val); if (bufsize != azx_dev->bufsize || @@ -2265,7 +2268,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, synchronize_irq(chip->irq); gcap = azx_readw(chip, GCAP); - snd_printdd("chipset global capabilities = 0x%x\n", gcap); + snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); /* ATI chips seems buggy about 64bit DMA addresses */ if (chip->driver_type == AZX_DRIVER_ATI) @@ -2309,7 +2312,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); if (!chip->azx_dev) { - snd_printk(KERN_ERR "cannot malloc azx_dev\n"); + snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); goto errout; } -- cgit v1.2.3 From fa7979663190240b838ab8c8bad7f59e618bf77c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 May 2009 12:50:04 +0200 Subject: ALSA: hda - Fix digital beep tone calculation The digital beep tone is calculated in two different ways depending on the codec chip. The standard one is using a divider, and another one is a linear tone for IDT/STAC codecs. Currently, only the latter type is used for all codecs, which resulted in a wrong tone pitch. This patch adds the calculation of the standard HD-audio type. Also clean-up the fields in hda_beep struct. Reference: bko#13162 http://bugzilla.kernel.org/show_bug.cgi?id=13162 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 55 +++++++++++++++++++++++++++++++++--------- sound/pci/hda/hda_beep.h | 5 ++-- sound/pci/hda/patch_sigmatel.c | 2 ++ 3 files changed, 49 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 4de5bacd3929..29272f2e95a0 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -45,6 +45,46 @@ static void snd_hda_generate_beep(struct work_struct *work) AC_VERB_SET_BEEP_CONTROL, beep->tone); } +/* (non-standard) Linear beep tone calculation for IDT/STAC codecs + * + * The tone frequency of beep generator on IDT/STAC codecs is + * defined from the 8bit tone parameter, in Hz, + * freq = 48000 * (257 - tone) / 1024 + * that is from 12kHz to 93.75kHz in step of 46.875 hz + */ +static int beep_linear_tone(struct hda_beep *beep, int hz) +{ + hz *= 1000; /* fixed point */ + hz = hz - DIGBEEP_HZ_MIN; + if (hz < 0) + hz = 0; /* turn off PC beep*/ + else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) + hz = 0xff; + else { + hz /= DIGBEEP_HZ_STEP; + hz++; + } + return hz; +} + +/* HD-audio standard beep tone parameter calculation + * + * The tone frequency in Hz is calculated as + * freq = 48000 / (tone * 4) + * from 47Hz to 12kHz + */ +static int beep_standard_tone(struct hda_beep *beep, int hz) +{ + if (hz <= 0) + return 0; /* disabled */ + hz = 12000 / hz; + if (hz > 0xff) + return 0xff; + if (hz <= 0) + return 1; + return hz; +} + static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, unsigned int code, int hz) { @@ -55,21 +95,14 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, if (hz) hz = 1000; case SND_TONE: - hz *= 1000; /* fixed point */ - hz = hz - DIGBEEP_HZ_MIN; - if (hz < 0) - hz = 0; /* turn off PC beep*/ - else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) - hz = 0xff; - else { - hz /= DIGBEEP_HZ_STEP; - hz++; - } + if (beep->linear_tone) + beep->tone = beep_linear_tone(beep, hz); + else + beep->tone = beep_standard_tone(beep, hz); break; default: return -1; } - beep->tone = hz; /* schedule beep event */ schedule_work(&beep->beep_work); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 51bf6a5daf39..0c3de787c717 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -30,8 +30,9 @@ struct hda_beep { struct hda_codec *codec; char phys[32]; int tone; - int nid; - int enabled; + hda_nid_t nid; + unsigned int enabled:1; + unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ struct work_struct beep_work; /* scheduled task for beep event */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 02950980778e..17310814f121 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3737,6 +3737,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = snd_hda_attach_beep_device(codec, nid); if (err < 0) return err; + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = 1; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (codec->beep && -- cgit v1.2.3 From 4fcd39207f4c91185cc89e3e6a28cbb643034ff1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 May 2009 18:34:52 +0200 Subject: ALSA: hda - Reset CORB/RIRB at retrying the verb communication When a codec communication error occurs, the CORB/RIRB counters should be reset first before re-issuing the verb. Simply call azx_free_cmd_io() and azx_init_cmd_io() to achieve that. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 49fd973b85cc..3fc75e2061ab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -522,6 +522,7 @@ static void azx_init_cmd_io(struct azx *chip) /* RIRB set up */ chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + chip->rirb.wp = chip->rirb.rp = chip->rirb.cmds = 0; azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr)); @@ -533,7 +534,6 @@ static void azx_init_cmd_io(struct azx *chip) azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN); - chip->rirb.rp = chip->rirb.cmds = 0; } static void azx_free_cmd_io(struct azx *chip) @@ -654,9 +654,11 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) snd_printk(KERN_ERR SFX "azx_get_response timeout (ERROR): " "last cmd=0x%08x\n", chip->last_cmd); + /* re-initialize CORB/RIRB */ spin_lock_irq(&chip->reg_lock); - chip->rirb.cmds = 0; /* reset the index */ bus->rirb_error = 1; + azx_free_cmd_io(chip); + azx_init_cmd_io(chip); spin_unlock_irq(&chip->reg_lock); return -1; } -- cgit v1.2.3 From 86d190e77c44cb057742dcc871b12ebd4633c387 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 15:18:58 +0200 Subject: ALSA: hda - Minor clean up of patch_sigmatel.c - Remove unneeded semicolons - Introduce spec->gpio_led to specify the GPIO bit for LED control Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 48 ++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 17310814f121..26d8707173b2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -193,6 +193,7 @@ struct sigmatel_spec { unsigned int gpio_dir; unsigned int gpio_data; unsigned int gpio_mute; + unsigned int gpio_led; /* stream */ unsigned int stream_delay; @@ -4651,22 +4652,13 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio_bit = 0; /* gets rid of compiler warning */ - - switch (spec->board_config) { - case STAC_HP_DV4_1222NR: - gpio_bit = 0x01; - break; - case STAC_HP_HDX: - gpio_bit = 0x08; - } if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data &= ~gpio_bit; /* orange */ + spec->gpio_data |= spec->gpio_led; /* white */ else - spec->gpio_data |= gpio_bit; /* white */ + spec->gpio_data &= ~spec->gpio_led; /* orange */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, @@ -5352,14 +5344,7 @@ again: */ spec->num_smuxes = 1; spec->num_dmuxes = 1; -#ifdef CONFIG_SND_HDA_POWER_SAVE - /* This controls MUTE LED */ - spec->gpio_mask |= 0x01; - spec->gpio_dir |= 0x01; - spec->gpio_data |= 0x01; - codec->patch_ops.check_power_status = - stac92xx_hp_check_power_status; -#endif + spec->gpio_led = 0x01; /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); @@ -5369,22 +5354,21 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; - /* - * For controlling MUTE LED on HP HDX16/HDX18 notebooks, - * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. - */ -#ifdef CONFIG_SND_HDA_POWER_SAVE /* orange/white mute led on GPIO3, orange=0, white=1 */ - spec->gpio_mask |= 0x08; - spec->gpio_dir |= 0x08; - spec->gpio_data |= 0x08; /* set to white */ + spec->gpio_led = 0x08; + break; + } +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (spec->gpio_led) { + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + spec->gpio_data |= spec->gpio_led; /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_check_power_status; + stac92xx_hp_check_power_status; + } #endif - break; - }; spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) @@ -5409,7 +5393,7 @@ again: codec->proc_widget_hook = stac92hd7x_proc_hook; return 0; -}; +} static int patch_stac922x(struct hda_codec *codec) { @@ -5564,7 +5548,7 @@ static int patch_stac927x(struct hda_codec *codec) /* correct the device field to SPDIF out */ snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070); break; - }; + } /* configure the analog microphone on some laptops */ snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ -- cgit v1.2.3 From 8174086167d43d0fd7b21928074145ae1d15bbab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 15:22:00 +0200 Subject: ALSA: hda - Allow concurrent RIRB access in single_cmd mode In the single_cmd mode, the current driver code doesn't do any update for RIRB just for any safety reason. But, actually the RIRB and single_cmd mode don't conflict. Unsolicited events can be delivered even while using the single_cmd mode. This patch allows the handling of unsolicited events with single_cmd mode, just always checking RIRB independent from single_cmd flag. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 82d5218a0125..01d8d97dca4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -901,8 +901,7 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - if (!chip->single_cmd) - azx_init_cmd_io(chip); + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); @@ -1018,7 +1017,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { - if (!chip->single_cmd && (status & RIRB_INT_RESPONSE)) + if (status & RIRB_INT_RESPONSE) azx_update_rirb(chip); azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } @@ -2338,11 +2337,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } /* allocate CORB/RIRB */ - if (!chip->single_cmd) { - err = azx_alloc_cmd_io(chip); - if (err < 0) - goto errout; - } + err = azx_alloc_cmd_io(chip); + if (err < 0) + goto errout; /* initialize streams */ azx_init_stream(chip); -- cgit v1.2.3 From aa2936f5fe060e95ae06685149645b234085a468 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 16:07:57 +0200 Subject: ALSA: hda - Support sync after writing a verb This patch adds a debug mode to make the codec communication synchronous. Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c, and the call of snd_hda_codec_write*() will become synchronous, i.e. wait for the reply from the codec at each time issuing a verb. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 113 +++++++++++++++++++++++++--------------------- 1 file changed, 61 insertions(+), 52 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 77385de01744..d1d5fb9d7afb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -158,6 +158,38 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, return val; } +/* + * Send and receive a verb + */ +static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, + unsigned int *res) +{ + struct hda_bus *bus = codec->bus; + int err, repeated = 0; + + if (res) + *res = -1; + snd_hda_power_up(codec); + mutex_lock(&bus->cmd_mutex); + again: + err = bus->ops.command(bus, cmd); + if (!err) { + if (res) { + *res = bus->ops.get_response(bus); + if (*res == -1 && bus->rirb_error) { + if (repeated++ < 1) { + snd_printd(KERN_WARNING "hda_codec: " + "Trying verb 0x%08x again\n", cmd); + goto again; + } + } + } + } + mutex_unlock(&bus->cmd_mutex); + snd_hda_power_down(codec); + return err; +} + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec @@ -174,31 +206,18 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { - struct hda_bus *bus = codec->bus; - unsigned int cmd, res; - int repeated = 0; - - cmd = make_codec_cmd(codec, nid, direct, verb, parm); - snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); - again: - if (!bus->ops.command(bus, cmd)) { - res = bus->ops.get_response(bus); - if (res == -1 && bus->rirb_error) { - if (repeated++ < 1) { - snd_printd(KERN_WARNING "hda_codec: " - "Trying verb 0x%08x again\n", cmd); - goto again; - } - } - } else - res = (unsigned int)-1; - mutex_unlock(&bus->cmd_mutex); - snd_hda_power_down(codec); + unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int res; + codec_exec_verb(codec, cmd, &res); return res; } EXPORT_SYMBOL_HDA(snd_hda_codec_read); +/* Define the below to send and receive verbs synchronously. + * If you often get any codec communication errors, this is worth to try. + */ +/* #define SND_HDA_SUPPORT_SYNC_WRITE */ + /** * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec @@ -214,17 +233,13 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { - struct hda_bus *bus = codec->bus; + unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); +#ifdef SND_HDA_SUPPORT_SYNC_WRITE unsigned int res; - int err; - - res = make_codec_cmd(codec, nid, direct, verb, parm); - snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); - err = bus->ops.command(bus, res); - mutex_unlock(&bus->cmd_mutex); - snd_hda_power_down(codec); - return err; + return codec_exec_verb(codec, cmd, &res); +#else + return codec_exec_verb(codec, cmd, NULL); +#endif } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -2281,28 +2296,22 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { - struct hda_bus *bus = codec->bus; - unsigned int res; - int err; + int err = snd_hda_codec_write(codec, nid, direct, verb, parm); + struct hda_cache_head *c; + u32 key; - res = make_codec_cmd(codec, nid, direct, verb, parm); - snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); - err = bus->ops.command(bus, res); - if (!err) { - struct hda_cache_head *c; - u32 key; - /* parm may contain the verb stuff for get/set amp */ - verb = verb | (parm >> 8); - parm &= 0xff; - key = build_cmd_cache_key(nid, verb); - c = get_alloc_hash(&codec->cmd_cache, key); - if (c) - c->val = parm; - } - mutex_unlock(&bus->cmd_mutex); - snd_hda_power_down(codec); - return err; + if (err < 0) + return err; + /* parm may contain the verb stuff for get/set amp */ + verb = verb | (parm >> 8); + parm &= 0xff; + key = build_cmd_cache_key(nid, verb); + mutex_lock(&codec->bus->cmd_mutex); + c = get_alloc_hash(&codec->cmd_cache, key); + if (c) + c->val = parm; + mutex_unlock(&codec->bus->cmd_mutex); + return 0; } EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); -- cgit v1.2.3 From b05a7d4fed7e51dca37d0a31baf1466de30b1f01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 May 2009 11:59:12 +0200 Subject: ALSA: hda - Always sync writes in single_cmd mode In the single_cmd mode, the hardware cannot store the multiple replies like on RIRB, thus each verb has to sync and wait for the response no matter whether the return value is needed or not. Otherwise it may result in a wrong return value from the previous verb. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 01d8d97dca4f..31a695e6e37d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -673,6 +673,27 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) * I left the codes, however, for debugging/testing purposes. */ +/* receive a response */ +static int azx_single_wait_for_response(struct azx *chip) +{ + int timeout = 50; + + while (timeout--) { + /* check IRV busy bit */ + if (azx_readw(chip, IRS) & ICH6_IRS_VALID) { + /* reuse rirb.res as the response return value */ + chip->rirb.res = azx_readl(chip, IR); + return 0; + } + udelay(1); + } + if (printk_ratelimit()) + snd_printd(SFX "get_response timeout: IRS=0x%x\n", + azx_readw(chip, IRS)); + chip->rirb.res = -1; + return -EIO; +} + /* send a command */ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) { @@ -688,7 +709,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) azx_writel(chip, IC, val); azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_BUSY); - return 0; + return azx_single_wait_for_response(chip); } udelay(1); } @@ -702,18 +723,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) static unsigned int azx_single_get_response(struct hda_bus *bus) { struct azx *chip = bus->private_data; - int timeout = 50; - - while (timeout--) { - /* check IRV busy bit */ - if (azx_readw(chip, IRS) & ICH6_IRS_VALID) - return azx_readl(chip, IR); - udelay(1); - } - if (printk_ratelimit()) - snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); - return (unsigned int)-1; + return chip->rirb.res; } /* -- cgit v1.2.3 From b21fadb9c1852c91622ca1dccfeb144bc535e36e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 May 2009 12:26:15 +0200 Subject: ALSA: hda - Add more register bits definitions Added some missing register bits definitions to reduce magic numbers. Also renamed some to follow the names on the datasheet. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 43 ++++++++++++++++++++++++++----------------- 1 file changed, 26 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 31a695e6e37d..f63bc6510e0f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -138,14 +138,23 @@ MODULE_DESCRIPTION("Intel HDA driver"); * registers */ #define ICH6_REG_GCAP 0x00 +#define ICH6_GCAP_64OK (1 << 0) /* 64bit address support */ +#define ICH6_GCAP_NSDO (3 << 1) /* # of serial data out signals */ +#define ICH6_GCAP_BSS (31 << 3) /* # of bidirectional streams */ +#define ICH6_GCAP_ISS (15 << 8) /* # of input streams */ +#define ICH6_GCAP_OSS (15 << 12) /* # of output streams */ #define ICH6_REG_VMIN 0x02 #define ICH6_REG_VMAJ 0x03 #define ICH6_REG_OUTPAY 0x04 #define ICH6_REG_INPAY 0x06 #define ICH6_REG_GCTL 0x08 +#define ICH6_GCTL_RESET (1 << 1) /* controller reset */ +#define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */ +#define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ #define ICH6_REG_WAKEEN 0x0c #define ICH6_REG_STATESTS 0x0e #define ICH6_REG_GSTS 0x10 +#define ICH6_GSTS_FSTS (1 << 1) /* flush status */ #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 #define ICH6_REG_WALCLK 0x30 @@ -153,17 +162,27 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 #define ICH6_REG_CORBWP 0x48 -#define ICH6_REG_CORBRP 0x4A +#define ICH6_REG_CORBRP 0x4a +#define ICH6_CORBRP_RST (1 << 15) /* read pointer reset */ #define ICH6_REG_CORBCTL 0x4c +#define ICH6_CORBCTL_RUN (1 << 1) /* enable DMA */ +#define ICH6_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ #define ICH6_REG_CORBSTS 0x4d +#define ICH6_CORBSTS_CMEI (1 << 0) /* memory error indication */ #define ICH6_REG_CORBSIZE 0x4e #define ICH6_REG_RIRBLBASE 0x50 #define ICH6_REG_RIRBUBASE 0x54 #define ICH6_REG_RIRBWP 0x58 +#define ICH6_RIRBWP_RST (1 << 15) /* write pointer reset */ #define ICH6_REG_RINTCNT 0x5a #define ICH6_REG_RIRBCTL 0x5c +#define ICH6_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ +#define ICH6_RBCTL_DMA_EN (1 << 1) /* enable DMA */ +#define ICH6_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ #define ICH6_REG_RIRBSTS 0x5d +#define ICH6_RBSTS_IRQ (1 << 0) /* response irq */ +#define ICH6_RBSTS_OVERRUN (1 << 2) /* overrun irq */ #define ICH6_REG_RIRBSIZE 0x5e #define ICH6_REG_IC 0x60 @@ -260,16 +279,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ #define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ -/* GCTL unsolicited response enable bit */ -#define ICH6_GCTL_UREN (1<<8) - -/* GCTL reset bit */ -#define ICH6_GCTL_RESET (1<<0) - -/* CORB/RIRB control, read/write pointer */ -#define ICH6_RBCTL_DMA_EN 0x02 /* enable DMA */ -#define ICH6_RBCTL_IRQ_EN 0x01 /* enable IRQ */ -#define ICH6_RBRWP_CLR 0x8000 /* read/write pointer clear */ /* below are so far hardcoded - should read registers in future */ #define ICH6_MAX_CORB_ENTRIES 256 #define ICH6_MAX_RIRB_ENTRIES 256 @@ -515,9 +524,9 @@ static void azx_init_cmd_io(struct azx *chip) /* set the corb write pointer to 0 */ azx_writew(chip, CORBWP, 0); /* reset the corb hw read pointer */ - azx_writew(chip, CORBRP, ICH6_RBRWP_CLR); + azx_writew(chip, CORBRP, ICH6_CORBRP_RST); /* enable corb dma */ - azx_writeb(chip, CORBCTL, ICH6_RBCTL_DMA_EN); + azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN); /* RIRB set up */ chip->rirb.addr = chip->rb.addr + 2048; @@ -529,7 +538,7 @@ static void azx_init_cmd_io(struct azx *chip) /* set the rirb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, RIRBSIZE, 0x02); /* reset the rirb hw write pointer */ - azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR); + azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST); /* set N=1, get RIRB response interrupt for new entry */ azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ @@ -796,7 +805,7 @@ static int azx_reset(struct azx *chip) } /* Accept unsolicited responses */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN); + azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL); /* detect codecs */ if (!chip->codec_mask) { @@ -2284,10 +2293,10 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, /* ATI chips seems buggy about 64bit DMA addresses */ if (chip->driver_type == AZX_DRIVER_ATI) - gcap &= ~0x01; + gcap &= ~ICH6_GCAP_64OK; /* allow 64bit DMA address if supported by H/W */ - if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) + if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); else { pci_set_dma_mask(pci, DMA_BIT_MASK(32)); -- cgit v1.2.3 From ba84bfcd2b6fdc5a9ac53a4ab103088c99f84a12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 30 May 2009 08:59:03 +0200 Subject: ALSA: hda - Fix reverted LED setup for HP The commit 86d190e77c44cb057742dcc871b12ebd4633c387 reverted the bit flip of LED GPIO for HP DX and DV4-1222nr. Fixed now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a1b4c9496d47..a915f404f654 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4666,9 +4666,9 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data |= spec->gpio_led; /* white */ - else spec->gpio_data &= ~spec->gpio_led; /* orange */ + else + spec->gpio_data |= spec->gpio_led; /* white */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, -- cgit v1.2.3 From 8a933ece41a59ce077eeffe5b9bf08b14d173c58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 31 May 2009 09:28:12 +0200 Subject: ALSA: hda - Fix a typo in the previous patch ICH6_GCTL_RESET was wrongly set to another bit by the commit b21fadb9c1852c91622ca1dccfeb144bc535e36e. This caused a problem when the codec needs really a reset (e.g. recovering from the communication error at probe). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f63bc6510e0f..b063d0e3d325 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -148,7 +148,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_OUTPAY 0x04 #define ICH6_REG_INPAY 0x06 #define ICH6_REG_GCTL 0x08 -#define ICH6_GCTL_RESET (1 << 1) /* controller reset */ +#define ICH6_GCTL_RESET (1 << 0) /* controller reset */ #define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */ #define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ #define ICH6_REG_WAKEEN 0x0c -- cgit v1.2.3 From a3d6ab9723060e8d67e28a8d9142e6d08953d07b Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Mon, 1 Jun 2009 16:37:28 +0800 Subject: ALSA: hda - Support NVIDIA 8 channel HDMI audio Support 8 channel HDMI audio for MCP78/7A Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 279 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 248 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index d57d8132a06e..f5792e2eea82 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -35,9 +35,28 @@ struct nvhdmi_spec { struct hda_pcm pcm_rec; }; +#define Nv_VERB_SET_Channel_Allocation 0xF79 +#define Nv_VERB_SET_Info_Frame_Checksum 0xF7A +#define Nv_VERB_SET_Audio_Protection_On 0xF98 +#define Nv_VERB_SET_Audio_Protection_Off 0xF99 + +#define Nv_Master_Convert_nid 0x04 +#define Nv_Master_Pin_nid 0x05 + +static hda_nid_t nvhdmi_convert_nids[4] = { + /*front, rear, clfe, rear_surr */ + 0x6, 0x8, 0xa, 0xc, +}; + static struct hda_verb nvhdmi_basic_init[] = { + /* set audio protect on */ + { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, /* enable digital output on pin widget */ - { 0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0x7, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0x9, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0xb, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0xd, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, {} /* terminator */ }; @@ -66,48 +85,205 @@ static int nvhdmi_init(struct hda_codec *codec) * Digital out */ static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -static int nvhdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int nvhdmi_dig_playback_pcm_close_8ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; + int i; + + snd_hda_codec_write(codec, Nv_Master_Convert_nid, + 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + for (i = 0; i < 4; i++) { + /* set the stream id */ + snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0, + AC_VERB_SET_CHANNEL_STREAMID, 0); + /* set the stream format */ + snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0, + AC_VERB_SET_STREAM_FORMAT, 0); + } + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } -static int nvhdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int nvhdmi_dig_playback_pcm_close_2ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct nvhdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + int chs; + unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + int i; + + mutex_lock(&codec->spdif_mutex); + + chs = substream->runtime->channels; + chan = chs ? (chs - 1) : 1; + + switch (chs) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; + dataDCC2 = 0x2; + + /* set the Audio InforFrame Channel Allocation */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write(codec, + Nv_Master_Convert_nid, + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + + /* set the stream id */ + snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | 0x0); + + /* set the stream format */ + snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); + + /* turn on again (if needed) */ + /* enable and set the channel status audio/data flag */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { + snd_hda_codec_write(codec, + Nv_Master_Convert_nid, + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + snd_hda_codec_write(codec, + Nv_Master_Convert_nid, + 0, + AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); + } + + for (i = 0; i < 4; i++) { + if (chs == 2) + channel_id = 0; + else + channel_id = i * 2; + + /* turn off SPDIF once; + *otherwise the IEC958 bits won't be updated + */ + if (codec->spdif_status_reset && + (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + /* set the stream id */ + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_CHANNEL_STREAMID, + (stream_tag << 4) | channel_id); + /* set the stream format */ + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_STREAM_FORMAT, + format); + /* turn on again (if needed) */ + /* enable and set the channel status audio/data flag */ + if (codec->spdif_status_reset && + (codec->spdif_ctls & AC_DIG1_ENABLE)) { + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); + } + } + + /* set the Audio Info Frame Checksum */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); + + mutex_unlock(&codec->spdif_mutex); + return 0; +} + +static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); + format, substream); } -static struct hda_pcm_stream nvhdmi_pcm_digital_playback = { +static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = Nv_Master_Convert_nid, + .rates = SNDRV_PCM_RATE_48000, + .maxbps = 16, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .ops = { + .open = nvhdmi_dig_playback_pcm_open, + .close = nvhdmi_dig_playback_pcm_close_8ch, + .prepare = nvhdmi_dig_playback_pcm_prepare_8ch + }, +}; + +static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .nid = 0x4, /* NID to query formats and rates and setup streams */ + .nid = Nv_Master_Convert_nid, .rates = SNDRV_PCM_RATE_48000, .maxbps = 16, .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = nvhdmi_dig_playback_pcm_open, - .close = nvhdmi_dig_playback_pcm_close, - .prepare = nvhdmi_dig_playback_pcm_prepare + .close = nvhdmi_dig_playback_pcm_close_2ch, + .prepare = nvhdmi_dig_playback_pcm_prepare_2ch }, }; -static int nvhdmi_build_pcms(struct hda_codec *codec) +static int nvhdmi_build_pcms_8ch(struct hda_codec *codec) { struct nvhdmi_spec *spec = codec->spec; struct hda_pcm *info = &spec->pcm_rec; @@ -117,7 +293,24 @@ static int nvhdmi_build_pcms(struct hda_codec *codec) info->name = "NVIDIA HDMI"; info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] + = nvhdmi_pcm_digital_playback_8ch; + + return 0; +} + +static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "NVIDIA HDMI"; + info->pcm_type = HDA_PCM_TYPE_HDMI; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] + = nvhdmi_pcm_digital_playback_2ch; return 0; } @@ -127,14 +320,40 @@ static void nvhdmi_free(struct hda_codec *codec) kfree(codec->spec); } -static struct hda_codec_ops nvhdmi_patch_ops = { +static struct hda_codec_ops nvhdmi_patch_ops_8ch = { + .build_controls = nvhdmi_build_controls, + .build_pcms = nvhdmi_build_pcms_8ch, + .init = nvhdmi_init, + .free = nvhdmi_free, +}; + +static struct hda_codec_ops nvhdmi_patch_ops_2ch = { .build_controls = nvhdmi_build_controls, - .build_pcms = nvhdmi_build_pcms, + .build_pcms = nvhdmi_build_pcms_2ch, .init = nvhdmi_init, .free = nvhdmi_free, }; -static int patch_nvhdmi(struct hda_codec *codec) +static int patch_nvhdmi_8ch(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 8; + spec->multiout.dig_out_nid = Nv_Master_Convert_nid; + + codec->patch_ops = nvhdmi_patch_ops_8ch; + + return 0; +} + +static int patch_nvhdmi_2ch(struct hda_codec *codec) { struct nvhdmi_spec *spec; @@ -144,13 +363,11 @@ static int patch_nvhdmi(struct hda_codec *codec) codec->spec = spec; - spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = 0x4; /* NID for copying analog to digital, - * seems to be unused in pure-digital - * case. */ + spec->multiout.dig_out_nid = Nv_Master_Convert_nid; - codec->patch_ops = nvhdmi_patch_ops; + codec->patch_ops = nvhdmi_patch_ops_2ch; return 0; } @@ -159,11 +376,11 @@ static int patch_nvhdmi(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; -- cgit v1.2.3 From 7c922de709b90badc19705e4f998e6d5b44c419b Mon Sep 17 00:00:00 2001 From: Nickolas Lloyd Date: Mon, 1 Jun 2009 11:12:29 +0200 Subject: ALSA: hda - Jack Mode changes for Sigmatel boards This patch changes Line In as Out Switch and Mic In as Out Switch to enums for consistency, and causes all mic and line in ports to be probed and controls to be added appropriately. Signed-off-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 219 ++++++++++++++++++++++++++--------------- 1 file changed, 140 insertions(+), 79 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a915f404f654..48f4a36c4813 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -670,62 +670,6 @@ static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) return vref; } -static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int new_vref; - unsigned int error; - - if (ucontrol->value.enumerated.item[0] == 0) - new_vref = AC_PINCTL_VREF_80; - else if (ucontrol->value.enumerated.item[0] == 1) - new_vref = AC_PINCTL_VREF_GRD; - else - new_vref = AC_PINCTL_VREF_HIZ; - - if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { - error = stac92xx_vref_set(codec, - kcontrol->private_value, new_vref); - return error; - } - - return 0; -} - -static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); - if (vref == AC_PINCTL_VREF_80) - ucontrol->value.enumerated.item[0] = 0; - else if (vref == AC_PINCTL_VREF_GRD) - ucontrol->value.enumerated.item[0] = 1; - else if (vref == AC_PINCTL_VREF_HIZ) - ucontrol->value.enumerated.item[0] = 2; - - return 0; -} - -static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static char *texts[] = { - "Mic In", "Line In", "Line Out" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 3; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= 3) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -2652,7 +2596,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } -static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid) +static unsigned int stac92xx_get_default_vref(struct hda_codec *codec, + hda_nid_t nid) { unsigned int pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; @@ -2706,15 +2651,108 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, return 1; } -#define stac92xx_io_switch_info snd_ctl_boolean_mono_info +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int i; + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = kcontrol->private_value; + + if (nid == spec->mic_switch || nid == spec->line_switch) + i = 3; + else + i = 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = i; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= i) + uinfo->value.enumerated.item = i-1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + unsigned int vref = stac92xx_vref_get(codec, nid); + + if (vref == stac92xx_get_default_vref(codec, nid)) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref = 0; + unsigned int error; + hda_nid_t nid = kcontrol->private_value; + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = stac92xx_get_default_vref(codec, nid); + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else if (ucontrol->value.enumerated.item[0] == 2) + new_vref = AC_PINCTL_VREF_HIZ; + else + return 0; + + if (new_vref != stac92xx_vref_get(codec, nid)) { + error = stac92xx_vref_set(codec, nid, new_vref); + return error; + } + + return 0; +} + +static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2]; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + if (kcontrol->private_value == spec->line_switch) + texts[0] = "Line In"; + else + texts[0] = "Mic In"; + texts[1] = "Line Out"; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 2; + uinfo->count = 1; + + if (uinfo->value.enumerated.item >= 2) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - int io_idx = kcontrol-> private_value & 0xff; + hda_nid_t nid = kcontrol->private_value; + int io_idx = (nid == spec->mic_switch) ? 1 : 0; - ucontrol->value.integer.value[0] = spec->io_switch[io_idx]; + ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx]; return 0; } @@ -2722,9 +2760,9 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value >> 8; - int io_idx = kcontrol-> private_value & 0xff; - unsigned short val = !!ucontrol->value.integer.value[0]; + hda_nid_t nid = kcontrol->private_value; + int io_idx = (nid == spec->mic_switch) ? 1 : 0; + unsigned short val = !!ucontrol->value.enumerated.item[0]; spec->io_switch[io_idx] = val; @@ -2733,7 +2771,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ else { unsigned int pinctl = AC_PINCTL_IN_EN; if (io_idx) /* set VREF for mic */ - pinctl |= stac92xx_get_vref(codec, nid); + pinctl |= stac92xx_get_default_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } @@ -2891,6 +2929,34 @@ static struct snd_kcontrol_new stac_input_src_temp = { .put = stac92xx_mux_enum_put, }; +static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, + hda_nid_t nid, int idx) +{ + int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int control = 0; + struct sigmatel_spec *spec = codec->spec; + char name[22]; + + if (!((get_defcfg_connect(def_conf)) & AC_JACK_PORT_FIXED)) { + if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD + && nid == spec->line_switch) + control = STAC_CTL_WIDGET_IO_SWITCH; + else if (snd_hda_query_pin_caps(codec, nid) + & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT)) + control = STAC_CTL_WIDGET_DC_BIAS; + else if (nid == spec->mic_switch) + control = STAC_CTL_WIDGET_IO_SWITCH; + } + + if (control) { + strcpy(name, auto_pin_cfg_labels[idx]); + return stac92xx_add_control(codec->spec, control, + strcat(name, " Jack Mode"), nid); + } + + return 0; +} + static int stac92xx_add_input_source(struct sigmatel_spec *spec) { struct snd_kcontrol_new *knew; @@ -3253,7 +3319,9 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; int err; + int idx; err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, spec->multiout.dac_nids, @@ -3270,20 +3338,13 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } - if (spec->line_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Line In as Output Switch", - spec->line_switch << 8); - if (err < 0) - return err; - } - - if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, - "Mic Jack Mode", - spec->mic_switch); - if (err < 0) - return err; + for (idx = AUTO_PIN_MIC; idx <= AUTO_PIN_FRONT_LINE; idx++) { + nid = cfg->input_pins[idx]; + if (nid) { + err = stac92xx_add_jack_mode_control(codec, nid, idx); + if (err < 0) + return err; + } } return 0; @@ -4193,7 +4254,7 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int pinctl, conf; if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) { /* for mic pins, force to initialize */ - pinctl = stac92xx_get_vref(codec, nid); + pinctl = stac92xx_get_default_vref(codec, nid); pinctl |= AC_PINCTL_IN_EN; stac92xx_auto_set_pinctl(codec, nid, pinctl); } else { -- cgit v1.2.3 From 0e4835c198e7dd9d814de25a17a1d6682107c394 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 25 May 2009 16:44:20 +0200 Subject: ALSA: hda-intel: improve initialization for ALC262_HP_BPC model Fix issues for 3 generations of HP workstations. The modest modifications do the following: 1. Change the second MIC from device 3 to device 1 2. Init the "boost" values to "0" by default From: John Brown Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 41 ++++++++++++++++++++++++++++------------- 1 file changed, 28 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index add920a925c7..d0786e9c3310 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10639,31 +10639,46 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */ + /* Input mixer1: only unmute Mic */ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, /* Input mixer3 */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -- cgit v1.2.3 From 92b9de8342c6ad0d930333851190ed25b88b190c Mon Sep 17 00:00:00 2001 From: Kacper Szczesniak Date: Tue, 2 Jun 2009 00:55:19 +0200 Subject: ALSA: hda - Macbook[Pro] 5 6ch support this is a patch against current snapshot that adds: 6 channels support for the MB5 model Signed-off-by: Kacper Szczesniak Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 80 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 65 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0786e9c3310..12d6015e07fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6266,6 +6266,34 @@ static struct hda_channel_mode alc885_mbp_6ch_modes[2] = { { 6, alc885_mbp_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_mb5_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_mb5_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b @@ -6310,10 +6338,14 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = { }; static struct snd_kcontrol_new alc885_mb5_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("HP 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("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -6322,6 +6354,7 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT), { } /* end */ }; + static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6551,22 +6584,39 @@ static struct hda_verb alc882_macpro_init_verbs[] = { /* Macbook 5,1 */ static struct hda_verb alc885_mb5_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Front mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* LineOut mixer */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin: output 0 (0x0d) */ + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* HP Pin: output 0 (0x0c) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -6986,13 +7036,13 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc885_mbp3_init_hook, }, [ALC885_MB5] = { - .mixers = { alc885_mb5_mixer }, + .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mb5_init_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .channel_mode = alc885_mb5_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, -- cgit v1.2.3 From d22142aa1b23d64d01f87a94b5756ff6cbd1022f Mon Sep 17 00:00:00 2001 From: Ozan Çağlayan Date: Mon, 1 Jun 2009 23:13:17 +0300 Subject: ALSA: hda - fix audio on LG R510 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently, LG R510 is only able to produce sound on headphones, the internal speakers are not working. The user tested and confirmed that with model=Dell headphones, internal speakers and the microphone are working flawlessly. Tested-by: Serdar Soytetir Signed-off-by: Ozan Çağlayan Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12d6015e07fc..56c2f8460409 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12111,6 +12111,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- cgit v1.2.3 From 8871e5b91518a47284b6bc2603b44dbc79c85446 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 01:02:50 +0200 Subject: ALSA: hda - Reorder and clean-up ALC268 quirk table Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some entries using SND_PCI_QUIRK_MASK(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56c2f8460409..ad6cc4218049 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12111,17 +12111,16 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), - SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), - SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series", + ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), - SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff64, "TOSHIBA L305", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), + SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", + ALC268_TOSHIBA), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), - SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), {} }; -- cgit v1.2.3 From 8dd783304e6d0f7c2830365d63f75f08aa343e10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 01:16:07 +0200 Subject: ALSA: hda - Add codec bus reset and verb-retry at critical errors Some machines machine cause a severe CORB/RIRB stall in certain weird conditions, such as PA access at the start up together with fglrx driver. This seems unable to be recovered without the controller reset. This patch allows the bus controller reset at critical errors so that the communication gets recovered again. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 32 ++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 6 +++++- sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++++++++++------ 3 files changed, 55 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d1d5fb9d7afb..aa0e1c18b606 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -165,28 +165,29 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, unsigned int *res) { struct hda_bus *bus = codec->bus; - int err, repeated = 0; + int err; if (res) *res = -1; + again: snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); - again: err = bus->ops.command(bus, cmd); - if (!err) { - if (res) { - *res = bus->ops.get_response(bus); - if (*res == -1 && bus->rirb_error) { - if (repeated++ < 1) { - snd_printd(KERN_WARNING "hda_codec: " - "Trying verb 0x%08x again\n", cmd); - goto again; - } - } - } - } + if (!err && res) + *res = bus->ops.get_response(bus); mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); + if (res && *res == -1 && bus->rirb_error) { + if (bus->response_reset) { + snd_printd("hda_codec: resetting BUS due to " + "fatal communication error\n"); + bus->ops.bus_reset(bus); + } + goto again; + } + /* clear reset-flag when the communication gets recovered */ + if (!err) + bus->response_reset = 0; return err; } @@ -3894,11 +3895,10 @@ EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); /** * snd_hda_suspend - suspend the codecs * @bus: the HDA bus - * @state: suspsend state * * Returns 0 if successful. */ -int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) +int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c5bd40f77bb9..f5fa5d1f223c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -574,6 +574,8 @@ struct hda_bus_ops { /* attach a PCM stream */ int (*attach_pcm)(struct hda_bus *bus, struct hda_codec *codec, struct hda_pcm *pcm); + /* reset bus for retry verb */ + void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ void (*pm_notify)(struct hda_bus *bus); @@ -624,6 +626,8 @@ struct hda_bus { unsigned int needs_damn_long_delay :1; unsigned int shutdown :1; /* being unloaded */ unsigned int rirb_error:1; /* error in codec communication */ + unsigned int response_reset:1; /* controller was reset */ + unsigned int in_reset:1; /* during reset operation */ }; /* @@ -907,7 +911,7 @@ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); * power management */ #ifdef CONFIG_PM -int snd_hda_suspend(struct hda_bus *bus, pm_message_t state); +int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b063d0e3d325..44f9a0aa20c5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -661,14 +661,23 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) return -1; } - snd_printk(KERN_ERR SFX "azx_get_response timeout (ERROR): " - "last cmd=0x%08x\n", chip->last_cmd); - /* re-initialize CORB/RIRB */ - spin_lock_irq(&chip->reg_lock); + /* a fatal communication error; need either to reset or to fallback + * to the single_cmd mode + */ bus->rirb_error = 1; + if (!bus->response_reset && !bus->in_reset) { + bus->response_reset = 1; + return -1; /* give a chance to retry */ + } + + snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " + "switching to single_cmd mode: last cmd=0x%08x\n", + chip->last_cmd); + chip->single_cmd = 1; + bus->response_reset = 0; + /* re-initialize CORB/RIRB */ azx_free_cmd_io(chip); azx_init_cmd_io(chip); - spin_unlock_irq(&chip->reg_lock); return -1; } @@ -709,6 +718,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) struct azx *chip = bus->private_data; int timeout = 50; + bus->rirb_error = 0; while (timeout--) { /* check ICB busy bit */ if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) { @@ -1247,6 +1257,23 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, struct hda_pcm *cpcm); static void azx_stop_chip(struct azx *chip); +static void azx_bus_reset(struct hda_bus *bus) +{ + struct azx *chip = bus->private_data; + int i; + + bus->in_reset = 1; + azx_stop_chip(chip); + azx_init_chip(chip); + if (chip->initialized) { + for (i = 0; i < AZX_MAX_PCMS; i++) + snd_pcm_suspend_all(chip->pcm[i]); + snd_hda_suspend(chip->bus); + snd_hda_resume(chip->bus); + } + bus->in_reset = 0; +} + /* * Codec initialization */ @@ -1270,6 +1297,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, bus_temp.ops.command = azx_send_cmd; bus_temp.ops.get_response = azx_get_response; bus_temp.ops.attach_pcm = azx_attach_pcm_stream; + bus_temp.ops.bus_reset = azx_bus_reset; #ifdef CONFIG_SND_HDA_POWER_SAVE bus_temp.power_save = &power_save; bus_temp.ops.pm_notify = azx_power_notify; @@ -1997,7 +2025,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) - snd_hda_suspend(chip->bus, state); + snd_hda_suspend(chip->bus); azx_stop_chip(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); -- cgit v1.2.3 From b20f3b834673be9ead83a3c6f07fa3881d1a990f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 01:20:22 +0200 Subject: ALSA: hda - Limit codec-verb retry to limited hardwares The reset of a BUS controller during operations is somehow risky and shouldn't be done inevitably for devices that have apparently no such codec-communication problems. This patch adds the check of the hardware and limits the bus-reset capability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 12 ++---------- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/patch_sigmatel.c | 9 +++++++++ 4 files changed, 15 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index aa0e1c18b606..562403a23488 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -214,11 +214,6 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_read); -/* Define the below to send and receive verbs synchronously. - * If you often get any codec communication errors, this is worth to try. - */ -/* #define SND_HDA_SUPPORT_SYNC_WRITE */ - /** * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec @@ -235,12 +230,9 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); -#ifdef SND_HDA_SUPPORT_SYNC_WRITE unsigned int res; - return codec_exec_verb(codec, cmd, &res); -#else - return codec_exec_verb(codec, cmd, NULL); -#endif + return codec_exec_verb(codec, cmd, + codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index f5fa5d1f223c..cad79efaabc9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -624,6 +624,9 @@ struct hda_bus { /* misc op flags */ unsigned int needs_damn_long_delay :1; + unsigned int allow_bus_reset:1; /* allow bus reset at fatal error */ + unsigned int sync_write:1; /* sync after verb write */ + /* status for codec/controller */ unsigned int shutdown :1; /* being unloaded */ unsigned int rirb_error:1; /* error in codec communication */ unsigned int response_reset:1; /* controller was reset */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 44f9a0aa20c5..9f44645a1d04 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -665,7 +665,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) * to the single_cmd mode */ bus->rirb_error = 1; - if (!bus->response_reset && !bus->in_reset) { + if (bus->allow_bus_reset && !bus->response_reset && !bus->in_reset) { bus->response_reset = 1; return -1; /* give a chance to retry */ } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 48f4a36c4813..42f944bb641d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5375,6 +5375,15 @@ again: if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); + /* Some HP machines seem to have unstable codec communications + * especially with ATI fglrx driver. For recovering from the + * CORB/RIRB stall, allow the BUS reset and keep always sync + */ + if (spec->board_config == STAC_HP_DV5) { + codec->bus->sync_write = 1; + codec->bus->allow_bus_reset = 1; + } + spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- cgit v1.2.3 From 3b315d70b094e8b439358756a9084438fd7a71c2 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Tue, 2 Jun 2009 10:54:19 +0200 Subject: ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 99 ++++++++++++++++++++++++++-- 2 files changed, 95 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 818d06f54e36..29c6125a838a 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-8930g Acer Aspire 8930G medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ad6cc4218049..2bbb9e445cb1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -223,6 +223,7 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -313,6 +314,8 @@ struct alc_spec { const struct hda_channel_mode *channel_mode; int num_channel_mode; int need_dac_fix; + int const_channel_count; + int ext_channel_count; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -368,6 +371,7 @@ struct alc_config_preset { unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; int need_dac_fix; + int const_channel_count; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -462,7 +466,7 @@ static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - spec->multiout.max_channels); + spec->ext_channel_count); } static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, @@ -472,9 +476,12 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; + &spec->ext_channel_count); + if (err >= 0 && !spec->const_channel_count) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } return err; } @@ -854,8 +861,13 @@ static void setup_preset(struct alc_spec *spec, spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; - spec->multiout.max_channels = spec->channel_mode[0].channels; + if (preset->const_channel_count) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; @@ -1456,6 +1468,48 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { { } }; +/* + * ALC888 Acer Aspire 8930G model + */ + +static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal Front to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Internal Rear to Rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect Internal CLFE to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect HP out to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable all DACs */ +/* DAC DISABLE/MUTE 1? */ +/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, +/* DAC DISABLE/MUTE 2? */ +/* some bit here disables the other DACs. Init=0x4900 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, +/* Enable amplifiers */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } +}; + static struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { @@ -1508,6 +1562,17 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x1b; + alc_automute_amp(codec); +} + /* * ALC880 3-stack model * @@ -8758,6 +8823,7 @@ 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_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8790,6 +8856,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", @@ -9009,6 +9077,27 @@ 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_8930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_8930g_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_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_8930g_init_hook, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer }, -- cgit v1.2.3 From 018df41861475595a51d327b83fb5830462f7a53 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Thu, 4 Jun 2009 00:13:40 +0200 Subject: ALSA: hda - More Aspire 8930G fixes Enable all three capture channels, including the missing nid 7 which is the only one capable of capturing DMIC input Enable Headphone amp for the HP jack. This causes a volume boost for headphones, but does not cause any noticeable effect for light loads like other amps, so there is no need to make it configurable. Add Input Mix capture mux setting to capture the output of the playback input mux (that is, what goes out the speakers except for PCM) Hack another coef register because the stereo DMIC for some reason produces a nonstandard sum/difference signal. I found a bit to make it just use the sum signal for both channels, which makes it behave like a standard mono microphone. The stereo is useless anyway (they're 1cm apart). Tested working: Three capture channels, mic in, line in, DMIC. Tested not working: CD. Not sure why, might be unconnected in the actual hardware or a CD drive issue. Also looked at SPDIF. It appears to work (emitter lights up inside the HP out jack) but I lack a proper miniTOSLINK cable to test it. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 54 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2bbb9e445cb1..8699b7fb45b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1469,10 +1469,10 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { }; /* - * ALC888 Acer Aspire 8930G model + * ALC889 Acer Aspire 8930G model */ -static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Unselect Front Mic by default in input mixer 3 */ @@ -1492,7 +1492,7 @@ static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect HP out to Front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {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}, /* Enable all DACs */ @@ -1507,6 +1507,17 @@ static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { /* Enable amplifiers */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, +/* DMIC fix + * This laptop has a stereo digital microphone. The mics are only 1cm apart + * which makes the stereo useless. However, either the mic or the ALC889 + * makes the signal become a difference/sum signal instead of standard + * stereo, which is annoying. So instead we flip this bit which makes the + * codec replicate the sum signal to both channels, turning it into a + * normal mono mic. + */ +/* DMIC_CONTROL? Init value = 0x0001 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0003}, { } }; @@ -1531,6 +1542,38 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { } }; +static struct hda_input_mux alc889_capture_sources[3] = { + /* Digital mic only available on first "ADC" */ + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Front Mic", 0xb }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1562,7 +1605,7 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } -static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9081,22 +9124,22 @@ static struct alc_config_preset alc883_presets[] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_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, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .const_channel_count = 6, .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_2_capture_sources, + ARRAY_SIZE(alc889_capture_sources), + .input_mux = alc889_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_8930g_init_hook, + .init_hook = alc889_acer_aspire_8930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, -- cgit v1.2.3 From 65f759831179bb8922f2a91a989487161d476a94 Mon Sep 17 00:00:00 2001 From: Alexander Beregalov Date: Thu, 4 Jun 2009 13:46:16 +0400 Subject: ALSA: hda_intel: fix build error when !PM Fix this build error when CONFIG_PM is not set: ound/pci/hda/hda_intel.c: In function 'azx_bus_reset': sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all' sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend' sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume' Signed-off-by: Alexander Beregalov Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f44645a1d04..4e9ea7080270 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1260,17 +1260,20 @@ static void azx_stop_chip(struct azx *chip); static void azx_bus_reset(struct hda_bus *bus) { struct azx *chip = bus->private_data; - int i; bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip); +#ifdef CONFIG_PM if (chip->initialized) { + int i; + for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); } +#endif bus->in_reset = 0; } -- cgit v1.2.3 From 3e1647c5b54a91a7182e121cfe569e6f0bf167ec Mon Sep 17 00:00:00 2001 From: Guido Günther Date: Fri, 5 Jun 2009 00:47:26 +0200 Subject: ALSA: support Sony Vaio TT MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit with BIOS probing only we offer a non functional headphone swith and volume slider. Signed-off-by: Guido Günther Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 45 ++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 29c6125a838a..54b08053d029 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -159,6 +159,7 @@ ALC883/888 3stack-6ch-intel Intel DG33* boards asus-p5q ASUS P5Q-EM boards mb31 MacBook 3,1 + sony-vaio-tt Sony VAIO TT auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8699b7fb45b9..1ab1b92fdc1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -243,6 +243,7 @@ enum { ALC888_ASUS_EEE1601, ALC889A_MB31, ALC1200_ASUS_P5Q, + ALC883_SONY_VAIO_TT, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -8134,6 +8135,16 @@ static struct snd_kcontrol_new alc889A_mb31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_vaiott_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + static struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -8410,6 +8421,17 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; +static struct hda_verb alc883_vaiott_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static void alc888_3st_hp_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8669,6 +8691,16 @@ static void alc888_lenovo_sky_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc883_vaiott_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x17; + alc_automute_amp(codec); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -8884,6 +8916,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC889A_MB31] = "mb31", + [ALC883_SONY_VAIO_TT] = "sony-vaio-tt", [ALC883_AUTO] = "auto", }; @@ -8978,6 +9011,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), {} }; @@ -9373,6 +9407,17 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc889A_mb31_unsol_event, .init_hook = alc889A_mb31_automute, }, + [ALC883_SONY_VAIO_TT] = { + .mixers = { alc883_vaiott_mixer }, + .init_verbs = { alc883_init_verbs, alc883_vaiott_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_vaiott_init_hook, + }, }; -- cgit v1.2.3 From 64a8be74357477558183b43156c5536b642de134 Mon Sep 17 00:00:00 2001 From: David Heidelberger Date: Mon, 8 Jun 2009 16:15:18 +0200 Subject: ALSA: hda - Add 7.1 support for MSI GX620 Added 7.1 support for MSI GX620 and jack quirk. Reference: kernel bug#13430 http://bugzilla.kernel.org/show_bug.cgi?id=13430 Signed-off-by: David Heidelberger Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 +- sound/pci/hda/patch_realtek.c | 115 +++++++++++++++++++++++++-- 2 files changed, 109 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 54b08053d029..de8e10a94103 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -143,7 +143,8 @@ ALC883/888 medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI - targa-2ch-dig Targs/MSI with 2-channel + targa-2ch-dig Targa/MSI with 2-channel + targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ab1b92fdc1c..cfe4808c3a4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -220,6 +220,7 @@ enum { ALC883_6ST_DIG, ALC883_TARGA_DIG, ALC883_TARGA_2ch_DIG, + ALC883_TARGA_8ch_DIG, ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, @@ -2722,6 +2723,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = { /* Enable GPIO mask and set output */ #define alc880_gpio1_init_verbs alc_gpio1_init_verbs #define alc880_gpio2_init_verbs alc_gpio2_init_verbs +#define alc880_gpio3_init_verbs alc_gpio3_init_verbs /* Clevo m520g init */ static struct hda_verb alc880_pin_clevo_init_verbs[] = { @@ -7719,6 +7721,73 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { { 6, alc883_3ST_ch6_init }, }; + +/* + * 2ch mode + */ +static struct hda_verb alc883_4ST_ch2_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PhIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_4ST_ch4_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_4ST_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc883_4ST_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { + { 2, alc883_4ST_ch2_init }, + { 4, alc883_4ST_ch4_init }, + { 6, alc883_4ST_ch6_init }, + { 8, alc883_4ST_ch8_init }, +}; + + /* * 2ch mode */ @@ -8355,14 +8424,24 @@ static struct hda_verb alc883_tagra_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ +/* Connect Line-Out side jack (SPDIF) to Side */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, +/* Connect Mic jack to CLFE */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect Line-in jack to Surround */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect HP out jack to Front */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -8607,8 +8686,8 @@ static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); @@ -8895,6 +8974,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_6ST_DIG] = "6stack-dig", [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", + [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", @@ -8979,6 +9059,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), @@ -9108,6 +9189,24 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_tagra_unsol_event, .init_hook = alc883_tagra_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 }, + .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, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes), + .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_automute, + }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, /* On TravelMate laptops, GPIO 0 enables the internal speaker -- cgit v1.2.3 From f03ecf50534a81b06544c58a713076d59d54baf9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 16:38:17 +0200 Subject: ALSA: hda - Fix the previous tagra-8ch patch - Fix a typo in the patch - Adapted to follow the recent change for unsol event handling Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfe4808c3a4c..337d2a59c67e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7726,7 +7726,7 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { * 2ch mode */ static struct hda_verb alc883_4ST_ch2_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PhIN_OUT }, + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -9205,7 +9205,7 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_automute, + .init_hook = alc883_tagra_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, -- cgit v1.2.3