From 2bac647c45ed9b397ace9109e6935c1364690f8a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 May 2007 18:21:41 +0200 Subject: [ALSA] hda-codec - Add AD1884 / AD1984 codec support Added the support of AD1884 and AD1984 codec chips. Also experimental quirks for Thinkpad T61/X61 laptops with AD1984. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 345 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 342 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0e1a879663f..30248cd562c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1,7 +1,8 @@ /* - * HD audio interface patch for AD1981HD, AD1983, AD1986A, AD1988 + * HD audio interface patch for AD1884, AD1981HD, AD1983, AD1984, AD1986A, + * AD1988 * - * Copyright (c) 2005 Takashi Iwai + * Copyright (c) 2005-2007 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 @@ -61,7 +62,7 @@ struct ad198x_spec { int num_channel_mode; /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ + struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; @@ -2774,12 +2775,350 @@ static int patch_ad1988(struct hda_codec *codec) } +/* + * AD1884 / AD1984 + * + * port-B - front line/mic-in + * port-E - aux in/out + * port-F - aux in/out + * port-C - rear line/mic-in + * port-D - rear line/hp-out + * port-A - front line/hp-out + * + * AD1984 = AD1884 + two digital mic-ins + * + * FIXME: + * For simplicity, we share the single DAC for both HP and line-outs + * right now. The inidividual playbacks could be easily implemented, + * but no build-up framework is given, so far. + */ + +static hda_nid_t ad1884_dac_nids[1] = { + 0x04, +}; + +static hda_nid_t ad1884_adc_nids[2] = { + 0x08, 0x09, +}; + +static hda_nid_t ad1884_capsrc_nids[2] = { + 0x0c, 0x0d, +}; + +#define AD1884_SPDIF_OUT 0x02 + +static struct hda_input_mux ad1884_capture_source = { + .num_items = 4, + .items = { + { "Front Mic", 0x0 }, + { "Mic", 0x1 }, + { "CD", 0x2 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884_base_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), + /* + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), + */ + HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984_dmic_mixers[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, + HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, + HDA_OUTPUT), + { } /* end */ +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1884_init_verbs[] = { + /* DACs; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-A (HP) mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* HP selector - select DAC2 */ + {0x22, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-D (Line-out) mixer */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono selector */ + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C (rear mic) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + /* SPDIF output selector */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +static int patch_ad1884(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); + spec->multiout.dac_nids = ad1884_dac_nids; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids); + spec->adc_nids = ad1884_adc_nids; + spec->capsrc_nids = ad1884_capsrc_nids; + spec->input_mux = &ad1884_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1884_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1884_init_verbs; + spec->spdif_route = 0; + + codec->patch_ops = ad198x_patch_ops; + + return 0; +} + +/* + * Lenovo Thinkpad T61/X61 + */ +static struct hda_input_mux ad1984_thinkpad_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* additional verbs */ +static struct hda_verb ad1984_thinkpad_init_verbs[] = { + /* Port-E (docking station mic) pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* docking mic boost */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer - docking mic; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } /* end */ +}; + +/* Digial MIC ADC NID 0x05 + 0x06 */ +static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_setup_stream(codec, 0x05 + substream->number, + stream_tag, 0, format); + return 0; +} + +static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_setup_stream(codec, 0x05 + substream->number, + 0, 0, 0); + return 0; +} + +static struct hda_pcm_stream ad1984_pcm_dmic_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x05, + .ops = { + .prepare = ad1984_pcm_dmic_prepare, + .cleanup = ad1984_pcm_dmic_cleanup + }, +}; + +static int ad1984_build_pcms(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + struct hda_pcm *info; + int err; + + err = ad198x_build_pcms(codec); + if (err < 0) + return err; + + info = spec->pcm_rec + codec->num_pcms; + codec->num_pcms++; + info->name = "AD1984 Digital Mic"; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture; + return 0; +} + +/* models */ +enum { + AD1984_BASIC, + AD1984_THINKPAD, + AD1984_MODELS +}; + +static const char *ad1984_models[AD1984_MODELS] = { + [AD1984_BASIC] = "basic", + [AD1984_THINKPAD] = "thinkpad", +}; + +static struct snd_pci_quirk ad1984_cfg_tbl[] = { + /* Lenovo Thinkpad T61/X61 */ + SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + {} +}; + +static int patch_ad1984(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config, err; + + err = patch_ad1884(codec); + if (err < 0) + return err; + spec = codec->spec; + board_config = snd_hda_check_board_config(codec, AD1984_MODELS, + ad1984_models, ad1984_cfg_tbl); + switch (board_config) { + case AD1984_BASIC: + /* additional digital mics */ + spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers; + codec->patch_ops.build_pcms = ad1984_build_pcms; + break; + case AD1984_THINKPAD: + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984_thinkpad_capture_source; + spec->mixers[0] = ad1984_thinkpad_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; + break; + } + return 0; +} + + /* * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, + { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, -- cgit v1.2.3 From ea7cfcdfe6439cd98816da9d339a6bc9032d2084 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Sat, 19 May 2007 17:21:04 +0200 Subject: [ALSA] ice1724 - Add PCM Playback Switch to Revo 7.1 This patch adds the support of mute for front channels of M-Audio Revolution 7.1 (the DAC AK4381 features a mute bit). Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/revo.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 690ceb34064..d18a31e188a 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -186,7 +186,12 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, #define AK_DAC(xname,xch) { .name = xname, .num_channels = xch } static const struct snd_akm4xxx_dac_channel revo71_front[] = { - AK_DAC("PCM Playback Volume", 2) + { + .name = "PCM Playback Volume", + .num_channels = 2, + /* front channels DAC supports muting */ + .switch_name = "PCM Playback Switch", + }, }; static const struct snd_akm4xxx_dac_channel revo71_surround[] = { -- cgit v1.2.3 From 0dae0f83cd9956d0959f6d6de9f5391da6483274 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 May 2007 12:41:29 +0200 Subject: [ALSA] hda-codec - Add support of newer version of Intel iMac Added the pin configs for newer version of Intel iMac. The information provided by Ivan N. Zlatev . Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e3964fc4c40..3653f57b7f6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -64,6 +64,7 @@ enum { STAC_MACBOOK_PRO_V1, STAC_MACBOOK_PRO_V2, STAC_IMAC_INTEL, + STAC_IMAC_INTEL_20, STAC_922X_MODELS }; @@ -567,6 +568,12 @@ static unsigned int imac_intel_pin_configs[10] = { 0x400000fc, 0x400000fb, }; +static unsigned int imac_intel_20_pin_configs[10] = { + 0x0121E21F, 0x90A7012E, 0x9017E110, 0x400000FD, + 0x400000FE, 0x0181E020, 0x1145E230, 0x11C5E240, + 0x400000FC, 0x400000FB, +}; + static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, @@ -576,6 +583,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_MACBOOK_PRO_V1] = macbook_pro_v1_pin_configs, [STAC_MACBOOK_PRO_V2] = macbook_pro_v2_pin_configs, [STAC_IMAC_INTEL] = imac_intel_pin_configs, + [STAC_IMAC_INTEL_20] = imac_intel_20_pin_configs, }; static const char *stac922x_models[STAC_922X_MODELS] = { @@ -587,6 +595,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_MACBOOK_PRO_V1] = "macbook-pro-v1", [STAC_MACBOOK_PRO_V2] = "macbook-pro", [STAC_IMAC_INTEL] = "imac-intel", + [STAC_IMAC_INTEL_20] = "imac-intel-20", }; static struct snd_pci_quirk stac922x_cfg_tbl[] = { @@ -2037,6 +2046,9 @@ static int patch_stac922x(struct hda_codec *codec) case 0x106b0700: /* Intel-based iMac */ spec->board_config = STAC_IMAC_INTEL; break; + case 0x106b1700: /* Intel iMac 20 */ + spec->board_config = STAC_IMAC_INTEL_20; + break; } } -- cgit v1.2.3 From 0ba7962b9f06c02dd1af93002e8d757805d16758 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 May 2007 16:27:32 +0200 Subject: [ALSA] hda-codec - Fix wrong mixer controls for AD1984 thinkpad model Fixed the wrong mixer controls for AD1984 thinkpad model. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 30248cd562c..4659fdeec3d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2979,11 +2979,9 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), -- cgit v1.2.3 From 621887aee9c7b4b613c12b82b83df7e56877f303 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 May 2007 18:46:54 +0200 Subject: [ALSA] Add support for Cyrix/NatSemi Geode CS5530 (VSA1) Add support for Cyrix/NatSemi Geode SC5530 (VSA1). The driver is snd-cs5530. Signed-off-by Ash Willis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/Kconfig | 10 ++ sound/pci/Makefile | 2 + sound/pci/cs5530.c | 306 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 318 insertions(+) create mode 100644 sound/pci/cs5530.c (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 61e35ecc57b..fa17786e158 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -215,6 +215,16 @@ config SND_CS46XX_NEW_DSP This works better than the old code, so say Y. +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND && ISA_DMA_API + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. + config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on SND && X86 && !X86_64 diff --git a/sound/pci/Makefile b/sound/pci/Makefile index e06736da9ef..cd76e0293d0 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -12,6 +12,7 @@ snd-azt3328-objs := azt3328.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_AZT3328) += snd-azt3328.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c new file mode 100644 index 00000000000..240a0a46220 --- /dev/null +++ b/sound/pci/cs5530.c @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis + * (C) Copyright 2003 Red Hat Inc + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + -- cgit v1.2.3 From dda369cf2022fb767b158c4ba2fe2f1ca1fba29c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 May 2007 11:50:33 +0200 Subject: [ALSA] Fix SB-module dependency with PCI drivers A few PCI drivers like ALS4000 and CS5530 require the SB16-codes. This patch fixes / improves the dependency between SB modules and PCI drivers. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fa17786e158..c6b44102aa5 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -33,6 +33,7 @@ config SND_ALS4000 select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_SB_COMMON help Say Y here to include support for soundcards based on Avance Logic ALS4000 chips. @@ -218,7 +219,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on SND && ISA_DMA_API - select SND_SB16 + select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- cgit v1.2.3 From 65ff2357a7e40bf4b42858516861c3d0e19f512b Mon Sep 17 00:00:00 2001 From: Ash Willis Date: Tue, 29 May 2007 14:34:17 +0200 Subject: [ALSA] Disable debugging output for the ALS300 driver Disables debugging output in the ALS300 driver. Also contains a whitespace cleanup and a fix for a potential bug. Signed-off-by: Ash Willis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/als300.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8afcb98ca7b..48cc39b771d 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -733,7 +733,8 @@ static int __devinit snd_als300_create(struct snd_card *card, snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + err = snd_als300_ac97(chip); + if (err < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err; -- cgit v1.2.3 From 5d5d3bc3eddf2ad97b2cb090b92580e7fed6cee1 Mon Sep 17 00:00:00 2001 From: "Ivan N. Zlatev" Date: Tue, 29 May 2007 16:03:00 +0200 Subject: [ALSA] hda-codec - Fix pin configs for Intel Macs * adds the pinconfigs for all 5 Apple boards and 14 Subsystem IDs (support for possibly all iMac, Mac, MacMini etc etc) * adds 'intel-mac-v1' to v5 models which replace the current * reflects changes in Alsa-Configuration.txt Signed-off-by: Ivan N. Zlatev Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 97 ++++++++++++++++++++++++++++-------------- 1 file changed, 65 insertions(+), 32 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3653f57b7f6..dd005ed89f0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -59,6 +59,12 @@ enum { STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, + STAC_INTEL_MAC_V1, + STAC_INTEL_MAC_V2, + STAC_INTEL_MAC_V3, + STAC_INTEL_MAC_V4, + STAC_INTEL_MAC_V5, + /* for backward compitability */ STAC_MACMINI, STAC_MACBOOK, STAC_MACBOOK_PRO_V1, @@ -550,46 +556,64 @@ static unsigned int d945gtp5_pin_configs[10] = { 0x02a19320, 0x40000100, }; -static unsigned int macbook_pro_v1_pin_configs[10] = { - 0x0321e230, 0x03a1e020, 0x9017e110, 0x01014010, - 0x01a19021, 0x0381e021, 0x1345e240, 0x13c5e22e, - 0x02a19320, 0x400000fb +static unsigned int intel_mac_v1_pin_configs[10] = { + 0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240, + 0x400000fc, 0x400000fb, +}; + +static unsigned int intel_mac_v2_pin_configs[10] = { + 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa, + 0x400000fc, 0x400000fb, }; -static unsigned int macbook_pro_v2_pin_configs[10] = { - 0x0221401f, 0x90a70120, 0x01813024, 0x01014010, - 0x400000fd, 0x01016011, 0x1345e240, 0x13c5e22e, +static unsigned int intel_mac_v3_pin_configs[10] = { + 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int imac_intel_pin_configs[10] = { - 0x0121e230, 0x90a70120, 0x9017e110, 0x400000fe, - 0x400000fd, 0x0181e021, 0x1145e040, 0x400000fa, +static unsigned int intel_mac_v4_pin_configs[10] = { + 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, + 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int imac_intel_20_pin_configs[10] = { - 0x0121E21F, 0x90A7012E, 0x9017E110, 0x400000FD, - 0x400000FE, 0x0181E020, 0x1145E230, 0x11C5E240, - 0x400000FC, 0x400000FB, +static unsigned int intel_mac_v5_pin_configs[10] = { + 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, + 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, + 0x400000fc, 0x400000fb, }; static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, - [STAC_MACMINI] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK_PRO_V1] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK_PRO_V2] = macbook_pro_v2_pin_configs, - [STAC_IMAC_INTEL] = imac_intel_pin_configs, - [STAC_IMAC_INTEL_20] = imac_intel_20_pin_configs, + [STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs, + [STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs, + [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, + [STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs, + [STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs, + /* for backward compitability */ + [STAC_MACMINI] = intel_mac_v3_pin_configs, + [STAC_MACBOOK] = intel_mac_v5_pin_configs, + [STAC_MACBOOK_PRO_V1] = intel_mac_v3_pin_configs, + [STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs, + [STAC_IMAC_INTEL] = intel_mac_v2_pin_configs, + [STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs, }; static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", + [STAC_INTEL_MAC_V1] = "intel-mac-v1", + [STAC_INTEL_MAC_V2] = "intel-mac-v2", + [STAC_INTEL_MAC_V3] = "intel-mac-v3", + [STAC_INTEL_MAC_V4] = "intel-mac-v4", + [STAC_INTEL_MAC_V5] = "intel-mac-v5", + /* for backward compitability */ [STAC_MACMINI] = "macmini", [STAC_MACBOOK] = "macbook", [STAC_MACBOOK_PRO_V1] = "macbook-pro-v1", @@ -658,7 +682,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* other systems */ /* Apple Mac Mini (early 2006) */ SND_PCI_QUIRK(0x8384, 0x7680, - "Mac Mini", STAC_MACMINI), + "Mac Mini", STAC_INTEL_MAC_V3), {} /* terminator */ }; @@ -2027,27 +2051,36 @@ static int patch_stac922x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, stac922x_cfg_tbl); - if (spec->board_config == STAC_MACMINI) { + if (spec->board_config == STAC_INTEL_MAC_V3) { spec->gpio_mute = 1; /* Intel Macs have all same PCI SSID, so we need to check * codec SSID to distinguish the exact models */ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id); switch (codec->subsystem_id) { - case 0x106b0a00: /* MacBook First generatoin */ - spec->board_config = STAC_MACBOOK; + + case 0x106b0800: + spec->board_config = STAC_INTEL_MAC_V1; break; - case 0x106b0200: /* MacBook Pro first generation */ - spec->board_config = STAC_MACBOOK_PRO_V1; + case 0x106b0600: + case 0x106b0700: + spec->board_config = STAC_INTEL_MAC_V2; break; - case 0x106b1e00: /* MacBook Pro second generation */ - spec->board_config = STAC_MACBOOK_PRO_V2; + case 0x106b0e00: + case 0x106b0f00: + case 0x106b1600: + case 0x106b1700: + case 0x106b0200: + case 0x106b1e00: + spec->board_config = STAC_INTEL_MAC_V3; break; - case 0x106b0700: /* Intel-based iMac */ - spec->board_config = STAC_IMAC_INTEL; + case 0x106b1a00: + case 0x00000100: + spec->board_config = STAC_INTEL_MAC_V4; break; - case 0x106b1700: /* Intel iMac 20 */ - spec->board_config = STAC_IMAC_INTEL_20; + case 0x106b0a00: + case 0x106b2200: + spec->board_config = STAC_INTEL_MAC_V5; break; } } -- cgit v1.2.3 From 807a463603a2e014f36e88317aa5e07c15887fcd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 May 2007 19:01:37 +0200 Subject: [ALSA] hda-codec - Fix STAC922x capture boost level STAC922x provides the capture boost level up to 4, but actually it works only up to 2. Since the range of the mixer is automatically defined from amp-capability bits, we need to override the value beforehand. snd_hda_override_amp_caps() is introduced for this purpose. The function patch_stac922x() calls this for NID 0x12 (Mux Capture Volume). This should fix another recording problem on Intel Macs. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dd005ed89f0..addcb2a0ae6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2127,6 +2127,13 @@ static int patch_stac922x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; + /* Fix Mux capture level; max to 2 */ + snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, + (0 << AC_AMPCAP_OFFSET_SHIFT) | + (2 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + return 0; } -- cgit v1.2.3 From ba8c3c37db0d6bf5e6de25016975ebfb558d2aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 May 2007 12:42:31 +0200 Subject: [ALSA] ali5451 - Fix invalid type of codec->irq field The irq field of struct snd_ali shouldn't be unsigned since it's uninitialized value is -1. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ali5451/ali5451.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 41543a4933e..e2c8666e06a 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -239,7 +239,7 @@ struct snd_ali_image { struct snd_ali { - unsigned long irq; + int irq; unsigned long port; unsigned char revision; @@ -2343,7 +2343,7 @@ static int __devinit snd_ali_probe(struct pci_dev *pci, strcpy(card->driver, "ALI5451"); strcpy(card->shortname, "ALI 5451"); - sprintf(card->longname, "%s at 0x%lx, irq %li", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, codec->port, codec->irq); snd_ali_printk("register card.\n"); -- cgit v1.2.3 From 304e3bb7a15d839cef64b24a1bcc1230b2ae6fa6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 May 2007 12:46:21 +0200 Subject: [ALSA] rme9652 - Fix the hw_pointer check The negative check in hw_pointer callback doesn't work because the value is unsigned. Cast to int in the comparison to fix this. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/rme9652/rme9652.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index bd7dbd267ed..2de27405a0b 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -406,7 +406,7 @@ static snd_pcm_uframes_t rme9652_hw_pointer(struct snd_rme9652 *rme9652) } else if (!frag) return 0; offset -= rme9652->max_jitter; - if (offset < 0) + if ((int)offset < 0) offset += period_size * 2; } else { if (offset > period_size + rme9652->max_jitter) { -- cgit v1.2.3 From 3b194401a31b8df9a07a1850e3fe44fc7ba832a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Jun 2007 18:32:23 +0200 Subject: [ALSA] hda-codec - Fix Oops with AD1984 thinkpad model Fixed Oops with AD1984 thinkpad model. Also fixed the wrong init verbs for NID 0x03 and 0x04, which have apparently no mute bit. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4659fdeec3d..c1216470595 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2883,8 +2883,8 @@ static struct snd_kcontrol_new ad1984_dmic_mixers[] = { */ static struct hda_verb ad1884_init_verbs[] = { /* DACs; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Port-A (HP) mixer */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -2960,7 +2960,7 @@ static int patch_ad1884(struct hda_codec *codec) * Lenovo Thinkpad T61/X61 */ static struct hda_input_mux ad1984_thinkpad_capture_source = { - .num_items = 4, + .num_items = 3, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, -- cgit v1.2.3 From 538c49c41bf5a0a739ad2a5417ec25e060ee9d9e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Jun 2007 12:13:34 +0200 Subject: [ALSA] hda-codec - Fix AD1984 basic model Fix the amp direction of digital mic capture volume mixer, which resulted in -EINVAL. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index c1216470595..dff2e791788 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2872,9 +2872,9 @@ static struct snd_kcontrol_new ad1984_dmic_mixers[] = { HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, - HDA_OUTPUT), + HDA_INPUT), HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, - HDA_OUTPUT), + HDA_INPUT), { } /* end */ }; -- cgit v1.2.3 From 7d87de2db2213e6e9413532445b14c92dae42c85 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 5 Jun 2007 12:17:21 +0200 Subject: [ALSA] hda-codec - Add proper model for HP xw series Set the proper model for HP xw4550, xw4600, xw6600 and xw8600. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4776de93928..c92c97c6cbe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7592,8 +7592,12 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), -- cgit v1.2.3 From a361d84bfcd938208dea6c84aa19994b3d69e15d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 5 Jun 2007 12:30:55 +0200 Subject: [ALSA] hda-codec - Add support of ALC268 codec Added the support of new ALC268 codec chip. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 517 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 517 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c92c97c6cbe..49a0fa8ec67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -98,6 +98,13 @@ enum { ALC262_MODEL_LAST /* last tag */ }; +/* ALC268 models */ +enum { + ALC268_3ST, + ALC268_AUTO, + ALC268_MODEL_LAST /* last tag */ +}; + /* ALC861 models */ enum { ALC861_3ST, @@ -7803,6 +7810,515 @@ static int patch_alc262(struct hda_codec *codec) return 0; } +/* + * ALC268 channel source setting (2 channel) + */ +#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID +#define alc268_modes alc260_modes + +static hda_nid_t alc268_dac_nids[2] = { + /* front, hp */ + 0x02, 0x03 +}; + +static hda_nid_t alc268_adc_nids[2] = { + /* ADC0-1 */ + 0x08, 0x07 +}; + +static hda_nid_t alc268_adc_nids_alt[1] = { + /* ADC0 */ + 0x08 +}; + +static struct snd_kcontrol_new alc268_base_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc268_base_init_verbs[] = { + /* Unmute DAC0-1 and set vol = 0 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* + * Set up output mixers (0x0c - 0x0e) + */ + /* set vol=0 to output mixers */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, + + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1c, 14, 15, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + { } +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc268_volume_init_verbs[] = { + /* set output DAC */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* set PCBEEP vol = 0 */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))}, + + { } +}; + +#define alc268_mux_enum_info alc_mux_enum_info +#define alc268_mux_enum_get alc_mux_enum_get + +static int alc268_mux_enum_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; + const struct hda_input_mux *imux = spec->input_mux; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; + hda_nid_t nid = capture_mixers[adc_idx]; + unsigned int *cur_val = &spec->cur_mux[adc_idx]; + unsigned int i, idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (*cur_val == idx && !codec->in_resume) + return 0; + for (i = 0; i < imux->num_items; i++) { + unsigned int v = (i == idx) ? 0x7000 : 0x7080; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + v | (imux->items[i].index << 8)); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, + idx ); + } + *cur_val = idx; + return 1; +} + +static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc268_mux_enum_info, + .get = alc268_mux_enum_get, + .put = alc268_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc268_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc268_mux_enum_info, + .get = alc268_mux_enum_get, + .put = alc268_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux alc268_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x3 }, + }, +}; + +/* create input playback/capture controls for the given pin */ +static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, + const char *ctlname, int idx) +{ + char name[32]; + int err; + + sprintf(name, "%s Playback Volume", ctlname); + if (nid == 0x14) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x02, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (nid == 0x15) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x03, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + } else + return -1; + sprintf(name, "%s Playback Switch", ctlname); + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); + if (err < 0) + return err; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.num_dacs = 2; /* only use one dac */ + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.dac_nids[0] = 2; + spec->multiout.dac_nids[1] = 3; + + nid = cfg->line_out_pins[0]; + if (nid) + alc268_new_analog_output(spec, nid, "Front", 0); + + nid = cfg->speaker_pins[0]; + if (nid == 0x1d) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Speaker Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + nid = cfg->hp_pins[0]; + if (nid) + alc268_new_analog_output(spec, nid, "Headphone", 0); + + nid = cfg->line_out_pins[1] | cfg->line_out_pins[2]; + if (nid == 0x16) { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} + +/* create playback/capture controls for input pins */ +static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + struct hda_input_mux *imux = &spec->private_imux; + int i, idx1; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch(cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + default: + continue; + } + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; + } + return 0; +} + +static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0]; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; + unsigned int dac_vol1, dac_vol2; + + if (speaker_nid) { + snd_hda_codec_write(codec, speaker_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + } else { + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); + } + + dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */ + if (line_nid == 0x14) + dac_vol2 = AMP_OUT_ZERO; + else if (line_nid == 0x15) + dac_vol1 = AMP_OUT_ZERO; + if (hp_nid == 0x14) + dac_vol2 = AMP_OUT_ZERO; + else if (hp_nid == 0x15) + dac_vol1 = AMP_OUT_ZERO; + if (line_nid != 0x16 || hp_nid != 0x16 || + spec->autocfg.line_out_pins[1] != 0x16 || + spec->autocfg.line_out_pins[2] != 0x16) + dac_vol1 = dac_vol2 = AMP_OUT_ZERO; + + snd_hda_codec_write(codec, 0x02, 0, + AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1); + snd_hda_codec_write(codec, 0x03, 0, + AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2); +} + +/* pcm configuration: identiacal with ALC880 */ +#define alc268_pcm_analog_playback alc880_pcm_analog_playback +#define alc268_pcm_analog_capture alc880_pcm_analog_capture +#define alc268_pcm_digital_playback alc880_pcm_digital_playback + +/* + * BIOS auto configuration + */ +static int alc268_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc268_ignore[] = { 0 }; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc268_ignore); + if (err < 0) + return err; + if (!spec->autocfg.line_outs) + return 0; /* can't find valid BIOS pin config */ + + err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = 2; + + /* digital only support output */ + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs; + spec->num_mux_defs = 1; + spec->input_mux = &spec->private_imux; + + return 1; +} + +#define alc268_auto_init_multi_out alc882_auto_init_multi_out +#define alc268_auto_init_hp_out alc882_auto_init_hp_out +#define alc268_auto_init_analog_input alc882_auto_init_analog_input + +/* init callback for auto-configuration model -- overriding the default init */ +static void alc268_auto_init(struct hda_codec *codec) +{ + alc268_auto_init_multi_out(codec); + alc268_auto_init_hp_out(codec); + alc268_auto_init_mono_speaker_out(codec); + alc268_auto_init_analog_input(codec); +} + +/* + * configuration and preset + */ +static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC268_3ST] = "3stack", + [ALC268_AUTO] = "auto", +}; + +static struct snd_pci_quirk alc268_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), + {} +}; + +static struct alc_config_preset alc268_presets[] = { + [ALC268_3ST] = { + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .init_verbs = { alc268_base_init_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .dig_out_nid = ALC268_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + }, +}; + +static int patch_alc268(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST, + alc268_models, + 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"); + board_config = ALC268_AUTO; + } + + if (board_config == ALC268_AUTO) { + /* automatic parse from the BIOS config */ + err = alc268_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC268_3ST; + } + } + + if (board_config != ALC268_AUTO) + setup_preset(spec, &alc268_presets[board_config]); + + spec->stream_name_analog = "ALC268 Analog"; + spec->stream_analog_playback = &alc268_pcm_analog_playback; + spec->stream_analog_capture = &alc268_pcm_analog_capture; + + spec->stream_name_digital = "ALC268 Digital"; + spec->stream_digital_playback = &alc268_pcm_digital_playback; + + if (board_config == ALC268_AUTO) { + if (!spec->adc_nids && spec->input_mux) { + /* check whether NID 0x07 is valid */ + unsigned int wcap = get_wcaps(codec, 0x07); + + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc268_adc_nids_alt; + spec->num_adc_nids = + ARRAY_SIZE(alc268_adc_nids_alt); + spec->mixers[spec->num_mixers] = + alc268_capture_alt_mixer; + spec->num_mixers++; + } else { + spec->adc_nids = alc268_adc_nids; + spec->num_adc_nids = + ARRAY_SIZE(alc268_adc_nids); + spec->mixers[spec->num_mixers] = + alc268_capture_mixer; + spec->num_mixers++; + } + } + } + codec->patch_ops = alc_patch_ops; + if (board_config == ALC268_AUTO) + spec->init_hook = alc268_auto_init; + + return 0; +} + /* * ALC861 channel source setting (2/6 channel selection for 3-stack) */ @@ -10728,6 +11244,7 @@ static int patch_alc662(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, + { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- cgit v1.2.3 From 35739bb179386cf52c13d0779a42ecfe18dedf7e Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Wed, 6 Jun 2007 14:48:52 +0200 Subject: [ALSA] Yet another Uniwill laptop with ALC861 codec From: Dave Jones Rediffed version of the patch from .. http://bugzilla.kernel.org/show_bug.cgi?id=8016 that seems to be lingering for some time. Original patch by: Andy Shevchenko Signed-off-by: Dave Jones Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 49a0fa8ec67..f09fc6e72f4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9291,6 +9291,7 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), + SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), -- cgit v1.2.3 From 6f3bf657edf9c592cf9d92e07ab385bb0ede48c5 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 8 Jun 2007 11:39:35 +0200 Subject: [ALSA] hda-codec - Add quirk for another MSI laptop to alc883_cfg_tbl[] This hg commit adds the SSID of another MSI laptop to the alc883_cfg_tbl[] quirk list. Verified in #alsa/Freenode on Thu, Jun 7, 2007. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f09fc6e72f4..a31735d2835 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6388,6 +6388,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), -- cgit v1.2.3 From b959d1f83fc9cb707aef86ccb3f65e20f9572428 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jun 2007 12:25:25 +0200 Subject: [ALSA] hda-codec - Fix Thinkpad X61/T61 outputs Enable EAPD bit on Thinkpad X61/T61. The recent model requires it explicitly to make any sounds. In future, it should be toggled automatically on demand. Also, added a beep volume/switch. Some people like it although I don't understand why :) Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index dff2e791788..760c897cde7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2982,6 +2982,8 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3011,6 +3013,8 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog mixer - docking mic; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* enable EAPD bit */ + {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } /* end */ }; -- cgit v1.2.3 From 15cc4458c5c93bd5a616410815d5d165f0fe0900 Mon Sep 17 00:00:00 2001 From: Peer Chen Date: Fri, 8 Jun 2007 13:55:10 +0200 Subject: [ALSA] Add the MCP73/77 support to hda_intel driver Add the MCP73/MCP77 support to hda driver. The patch base on kernel 2.6.22-rc4 Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2fa281cbef9..74e5593a58e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1788,6 +1788,12 @@ static struct pci_device_id azx_ids[] = { { 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ { 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ { 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ + { 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ + { 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ + { 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- cgit v1.2.3 From 13d457094bc364e942884266036fd1b2ab74308b Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Mon, 11 Jun 2007 12:21:20 +0200 Subject: [ALSA] emu10k1 - EMU 1212 with 16 capture channels * adding 8 more 32-bit capture channels (total of 16) for emu1010 cards * adding some code comments and card details description Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/emu10k1_main.c | 52 +++++++++++++++++++++++++++ sound/pci/emu10k1/emufx.c | 78 ++++++++++++++++++++++++++++++++++++++-- sound/pci/emu10k1/emumixer.c | 16 +++++++++ sound/pci/emu10k1/emupcm.c | 39 ++++++++++++++------ 4 files changed, 172 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 4a9b59ad8ab..72367411ee6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -694,6 +694,37 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return 0; } +/* + * EMU-1010 - details found out from this driver, official MS Win drivers, + * testing the card: + * + * Audigy2 (aka Alice2): + * --------------------- + * * communication over PCI + * * conversion of 32-bit data coming over EMU32 links from HANA FPGA + * to 2 x 16-bit, using internal DSP instructions + * * slave mode, clock supplied by HANA + * * linked to HANA using: + * 32 x 32-bit serial EMU32 output channels + * 16 x EMU32 input channels + * (?) x I2S I/O channels (?) + * + * FPGA (aka HANA): + * --------------- + * * provides all (?) physical inputs and outputs of the card + * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.) + * * provides clock signal for the card and Alice2 + * * two crystals - for 44.1kHz and 48kHz multiples + * * provides internal routing of signal sources to signal destinations + * * inputs/outputs to Alice2 - see above + * + * Current status of the driver: + * ---------------------------- + * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz) + * * PCM device nb. 2: + * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops + * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops + */ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) { unsigned int i; @@ -850,6 +881,27 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1); + /* Pavel Hofman - setting defaults for 8 more capture channels + * Defaults only, users will set their own values anyways, let's + * just copy/paste. + */ + + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1); #endif #if 0 /* Original */ diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index c02012cccd8..7206c0fa06f 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1123,6 +1123,11 @@ snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF; } +/* + * Used for emu1010 - conversion from 32-bit capture inputs from HANA + * to 2 x 16-bit registers in audigy - their values are read via DMA. + * Conversion is performed by Audigy DSP instructions of FX8010. + */ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16( struct snd_emu10k1_fx8010_code *icode, u32 *ptr, int tmp, int bit_shifter16, @@ -1193,7 +1198,11 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP); #if 1 - /* PCM front Playback Volume (independent from stereo mix) */ + /* PCM front Playback Volume (independent from stereo mix) + * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31) + * where gpr contains attenuation from corresponding mixer control + * (snd_emu10k1_init_stereo_control) + */ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT)); snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100); @@ -1549,7 +1558,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->emu1010) { snd_printk("EMU inputs on\n"); - /* Capture 8 channels of S32_LE sound */ + /* Capture 16 (originally 8) channels of S32_LE sound */ /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ @@ -1560,6 +1569,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); /* Right ADC in 1 of 2 */ gpr_map[gpr++] = 0x00000000; + /* Delaying by one sample: instead of copying the input + * value A_P16VIN to output A_FXBUS2 as in the first channel, + * we use an auxiliary register, delaying the value by one + * sample + */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; @@ -1583,6 +1597,66 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr_map[gpr++] = 0x00000000; snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); + /* Pavel Hofman - we still have voices, A_FXBUS2s, and + * A_P16VINs available - + * let's add 8 more capture channels - total of 16 + */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x10)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x12)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x14)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x16)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x18)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1a)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1c)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1e)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), + A_C_00000000, A_C_00000000); #if 0 for (z = 4; z < 8; z++) { diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 4db6e1ca166..7b2c1dcc533 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -77,6 +77,10 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol, return 0; } +/* + * Items labels in enum mixer controls assigning source data to + * each destination + */ static char *emu1010_src_texts[] = { "Silence", "Dock Mic A", @@ -133,6 +137,9 @@ static char *emu1010_src_texts[] = { "DSP 31", }; +/* + * List of data sources available for each destination + */ static unsigned int emu1010_src_regs[] = { EMU_SRC_SILENCE,/* 0 */ EMU_SRC_DOCK_MIC_A1, /* 1 */ @@ -189,6 +196,10 @@ static unsigned int emu1010_src_regs[] = { EMU_SRC_ALICE_EMU32B+0xf, /* 52 */ }; +/* + * Data destinations - physical EMU outputs. + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_output_dst[] = { EMU_DST_DOCK_DAC1_LEFT1, /* 0 */ EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */ @@ -216,6 +227,11 @@ static unsigned int emu1010_output_dst[] = { EMU_DST_HANA_ADAT+7, /* 23 */ }; +/* + * Data destinations - HANA outputs going to Alice2 (audigy) for + * capture (EMU32 + I2S links) + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_input_dst[] = { EMU_DST_ALICE2_EMU32_0, EMU_DST_ALICE2_EMU32_1, diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index ab4f5df5241..eda5cb373de 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1233,24 +1233,26 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = runtime->hw.rate_max = 48000; spin_lock_irq(&emu->reg_lock); if (emu->card_capabilities->emu1010) { - /* TODO + /* Nb. of channels has been increased to 16 */ + /* TODO * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000 * rate_min = 44100, * rate_max = 192000, - * channels_min = 8, - * channels_max = 8, + * channels_min = 16, + * channels_max = 16, * Need to add mixer control to fix sample rate * - * There are 16 mono channels of 16bits each. + * There are 32 mono channels of 16bits each. * 24bit Audio uses 2x channels over 16bit * 96kHz uses 2x channels over 48kHz * 192kHz uses 4x channels over 48kHz - * So, for 48kHz 24bit, one has 8 channels - * for 96kHz 24bit, one has 4 channels - * for 192kHz 24bit, one has 2 channels + * So, for 48kHz 24bit, one has 16 channels + * for 96kHz 24bit, one has 8 channels + * for 192kHz 24bit, one has 4 channels + * */ #if 1 switch (emu->emu1010.internal_clock) { @@ -1258,13 +1260,15 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) /* For 44.1kHz */ runtime->hw.rates = SNDRV_PCM_RATE_44100; runtime->hw.rate_min = runtime->hw.rate_max = 44100; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; case 1: /* For 48kHz */ runtime->hw.rates = SNDRV_PCM_RATE_48000; runtime->hw.rate_min = runtime->hw.rate_max = 48000; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; }; #endif @@ -1282,7 +1286,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) #endif runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE; /* efx_voices_mask[0] is expected to be zero - * efx_voices_mask[1] is expected to have 16bits set + * efx_voices_mask[1] is expected to have 32bits set */ } else { runtime->hw.channels_min = runtime->hw.channels_max = 0; @@ -1787,11 +1791,24 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; - emu->efx_voices_mask[1] = 0xffff; + if (emu->card_capabilities->emu1010) + /* Pavel Hofman - 32 voices will be used for + * capture (write mode) - + * each bit = corresponding voice + */ + emu->efx_voices_mask[1] = 0xffffffff; + else + emu->efx_voices_mask[1] = 0xffff; } else { emu->efx_voices_mask[0] = 0xffff0000; emu->efx_voices_mask[1] = 0; } + /* For emu1010, the control has to set 32 upper bits (voices) + * out of the 64 bits (voices) to true for the 16-channels capture + * to work correctly. Correct A_FXWC2 initial value (0xffffffff) + * is already defined but the snd_emu10k1_pcm_efx_voices_mask + * control can override this register's value. + */ kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu); if (!kctl) return -ENOMEM; -- cgit v1.2.3 From 1c3973240b94a48766b7bfe743a4fcaad4eab3eb Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Mon, 11 Jun 2007 12:23:31 +0200 Subject: [ALSA] ali5451 warning fix sound/pci/ali5451/ali5451.c: In function 'snd_ali_prepare': sound/pci/ali5451/ali5451.c:716: warning: 'R2' may be used uninitialized in this function Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ali5451/ali5451.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index e2c8666e06a..05b4c869694 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -731,8 +731,7 @@ static void snd_ali_detect_spdif_rate(struct snd_ali *codec) return; } - count = 0; - while (count++ <= 50000) { + for (count = 0; count <= 50000; count++) { snd_ali_delay(codec, 6); bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1)); R2 = bval & 0x1F; -- cgit v1.2.3 From febe3375ea690a6cf544c33fa0fea1a06ff451ee Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 12 Jun 2007 11:27:46 +0200 Subject: [ALSA] hda-codec - Add HP Pavillion quirk to Realtek code This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a31735d2835..1950875b694 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6380,6 +6380,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), -- cgit v1.2.3 From a18519e1e4783628182743e6cca5f9dfa6a9e3bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jun 2007 12:58:23 +0200 Subject: [ALSA] hda-codec - Add model for Toshiba A135 Added model=lenovo for Toshiba A135 with ALC861VD codec. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1950875b694..d2cadb2038d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10006,6 +10006,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), {} }; -- cgit v1.2.3 From 8c87286fdcb67f796cc9477a9bc1a2f1e173d99f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2007 12:11:16 +0200 Subject: [ALSA] hda-codec - Fix ALC662 auto-configuration code The auto-configuration code didn't initialize the verbs correctly. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2cadb2038d..9c5c3cfa5ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11169,12 +11169,10 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - else if (err > 0) - /* hack - override the init verbs */ - spec->init_verbs[0] = alc662_auto_init_verbs; + spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; - return err; + return 1; } /* additional initialization for auto-configuration model */ @@ -11211,7 +11209,7 @@ static int patch_alc662(struct hda_codec *codec) if (err < 0) { alc_free(codec); return err; - } else if (err) { + } else if (!err) { printk(KERN_INFO "hda_codec: Cannot set up configuration " "from BIOS. Using base mode...\n"); -- cgit v1.2.3 From b69f3748400517d1023337b97456a0a1cdc74a79 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 19 Jun 2007 12:13:32 +0200 Subject: [ALSA] hda-codec - Add Fujitsu Siemens v3515 support to patch_conexant This patch adds support for the Fujitsu Seimens Amilo Pro v3515 notebook. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bef214bcddd..90ebc2c4988 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -802,6 +802,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_FUJITSU), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), {} }; -- cgit v1.2.3 From 76c08828709129bdce6c6a325e0342ba73f2618f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jun 2007 12:17:42 +0200 Subject: [ALSA] hda-codec - Add model dell for Dell XPS M1210 Added a new model 'dell' for Dell XPS M1210 with STAC922x codec chip. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index addcb2a0ae6..36423ca693e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -59,6 +59,7 @@ enum { STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, + STAC_922X_DELL, STAC_INTEL_MAC_V1, STAC_INTEL_MAC_V2, STAC_INTEL_MAC_V3, @@ -586,10 +587,17 @@ static unsigned int intel_mac_v5_pin_configs[10] = { 0x400000fc, 0x400000fb, }; +static unsigned int stac922x_dell_pin_configs[10] = { + 0x0221121e, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221122f, 0x03451340, 0x40c003f2, + 0x50a003f3, 0x405003f4 +}; + static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, + [STAC_922X_DELL] = stac922x_dell_pin_configs, [STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs, [STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs, [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, @@ -608,6 +616,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", + [STAC_922X_DELL] = "dell", [STAC_INTEL_MAC_V1] = "intel-mac-v1", [STAC_INTEL_MAC_V2] = "intel-mac-v2", [STAC_INTEL_MAC_V3] = "intel-mac-v3", @@ -683,6 +692,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* Apple Mac Mini (early 2006) */ SND_PCI_QUIRK(0x8384, 0x7680, "Mac Mini", STAC_INTEL_MAC_V3), + /* Dell */ + SND_PCI_QUIRK(0x1028, 0x01d7, "Dell XPS M1210", STAC_922X_DELL), + {} /* terminator */ }; -- cgit v1.2.3 From 92a22beb756c53985e8ba98d39f2c8fc834557a4 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 19 Jun 2007 16:48:28 +0200 Subject: [ALSA] hda: add eapd support to additional idt codecs Added support for EAPD on the 927x, and 9227-8 IDT HDA codecs. Enabling EAPD powers the internal speaker amp otherwise there is no sound on systems with an internal amp. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 31 ++++++++++++++++++------------- 1 file changed, 18 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 36423ca693e..270539a2bbc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -844,6 +844,21 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) } } +static void stac92xx_enable_eapd(struct hda_codec *codec) +{ + /* Configure GPIO0 as output */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x00000001); + /* Configure GPIO0 as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); + /* Assert GPIO0 high */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, 0x00000001); + /* Enable GPIO0 */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, 0x00000001); +} + /* * Analog playback callbacks */ @@ -2205,7 +2220,8 @@ static int patch_stac927x(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dac_nids; - + stac92xx_enable_eapd(codec); + err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { @@ -2273,18 +2289,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - - /* Configure GPIO0 as EAPD output */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + stac92xx_enable_eapd(codec); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { -- cgit v1.2.3 From e25c05f1d5cf4e332db88b15e9daa1cabd6e17e0 Mon Sep 17 00:00:00 2001 From: Jonathan Phenix Date: Tue, 19 Jun 2007 18:31:28 +0200 Subject: [ALSA] hda-codec - Output MFG information for HDA devices This patch output information about the MFG (Modem Function Group) to the /proc/asound/card?/codec#? file. This information is important to identify if a HDA device has a modem or not. Signed-off-by: Jonathan Phenix Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_proc.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e313e685f16..ac15066fd30 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -250,6 +250,12 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + + if (codec->mfg) + snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg); + else + snd_iprintf(buffer, "No Modem Function Group found\n"); + if (! codec->afg) return; snd_iprintf(buffer, "Default PCM:\n"); -- cgit v1.2.3 From 0ac8551ea08e655fc6f35874803f091a17dacf90 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jun 2007 15:46:13 +0200 Subject: [ALSA] hda-codec - Add AD1882 codec support Added AD1882 codec support. It has currently two models, 3stack and 6stack. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 287 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 285 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 760c897cde7..4d7f8d11ad7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1,6 +1,6 @@ /* - * HD audio interface patch for AD1884, AD1981HD, AD1983, AD1984, AD1986A, - * AD1988 + * HD audio interface patch for AD1882, AD1884, AD1981HD, AD1983, AD1984, + * AD1986A, AD1988 * * Copyright (c) 2005-2007 Takashi Iwai * @@ -3113,10 +3113,293 @@ static int patch_ad1984(struct hda_codec *codec) } +/* + * AD1882 + * + * port-A - front hp-out + * port-B - front mic-in + * port-C - rear line-in, shared surr-out (3stack) + * port-D - rear line-out + * port-E - rear mic-in, shared clfe-out (3stack) + * port-F - rear surr-out (6stack) + * port-G - rear clfe-out (6stack) + */ + +static hda_nid_t ad1882_dac_nids[3] = { + 0x04, 0x03, 0x05 +}; + +static hda_nid_t ad1882_adc_nids[2] = { + 0x08, 0x09, +}; + +static hda_nid_t ad1882_capsrc_nids[2] = { + 0x0c, 0x0d, +}; + +#define AD1882_SPDIF_OUT 0x02 + +/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */ +static struct hda_input_mux ad1882_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x1 }, + { "Mic", 0x4 }, + { "Line", 0x2 }, + { "CD", 0x3 }, + { "Mix", 0x7 }, + }, +}; + +static struct snd_kcontrol_new ad1882_base_mixers[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882_3stack_mixers[] = { + HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = ad198x_ch_mode_info, + .get = ad198x_ch_mode_get, + .put = ad198x_ch_mode_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882_6stack_mixers[] = { + HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb ad1882_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +static struct hda_verb ad1882_ch4_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +static struct hda_verb ad1882_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } /* end */ +}; + +static struct hda_channel_mode ad1882_modes[3] = { + { 2, ad1882_ch2_init }, + { 4, ad1882_ch4_init }, + { 6, ad1882_ch6_init }, +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1882_init_verbs[] = { + /* DACs; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-A (HP) mixer */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* HP selector - select DAC2 */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-D (Line-out) mixer */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-C (line-in) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-C mixer - mute as input */ + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Port-E (mic-in) pin */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-E mixer - mute as input */ + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Port-F (surround) */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-G (CLFE) */ + {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + /* SPDIF output selector */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +/* models */ +enum { + AD1882_3STACK, + AD1882_6STACK, + AD1882_MODELS +}; + +static const char *ad1882_models[AD1986A_MODELS] = { + [AD1882_3STACK] = "3stack", + [AD1882_6STACK] = "6stack", +}; + + +static int patch_ad1882(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 6; + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = ad1882_dac_nids; + spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); + spec->adc_nids = ad1882_adc_nids; + spec->capsrc_nids = ad1882_capsrc_nids; + spec->input_mux = &ad1882_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1882_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1882_init_verbs; + spec->spdif_route = 0; + + codec->patch_ops = ad198x_patch_ops; + + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, AD1882_MODELS, + ad1882_models, NULL); + switch (board_config) { + default: + case AD1882_3STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1882_3stack_mixers; + spec->channel_mode = ad1882_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1882_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + break; + case AD1882_6STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1882_6stack_mixers; + break; + } + return 0; +} + + /* * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, -- cgit v1.2.3 From e65365de5ba280e058bd6b8b80c8790253268887 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jun 2007 12:09:32 +0200 Subject: [ALSA] Fix invalid schedule_timeout_interruptible() Fixed the invalid use of schedule_timeout_interruptible() without checking pending signals. Simply replaced with schedule_timeout(). Suggestions thanks to Jeff Garzik. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 2 +- sound/pci/via82xx.c | 4 ++-- sound/pci/via82xx_modem.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 74e5593a58e..c78ff901a57 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -538,7 +538,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (! chip->rirb.cmds) return chip->rirb.res; /* the last value */ - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_after_eq(timeout, jiffies)); if (chip->msi) { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 50c9f92cfd1..6ea09df0c73 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2098,7 +2098,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; - schedule_timeout_uninterruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) @@ -2117,7 +2117,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) chip->ac97_secondary = 1; goto __ac97_ok2; } - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 8cbf8eba4ae..72425e73aba 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -983,7 +983,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; - schedule_timeout_uninterruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) @@ -1001,7 +1001,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) chip->ac97_secondary = 1; goto __ac97_ok2; } - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ -- cgit v1.2.3 From 6963f84c428ecd31780c97224004c2462b0664f7 Mon Sep 17 00:00:00 2001 From: Mike Crash Date: Mon, 25 Jun 2007 12:12:51 +0200 Subject: [ALSA] hda: support for S/PDIF out on ASUS M2V Added support for S/PDIF digital output on ASUS M2V motheboard - added new model '3stack-660-digout' and ALC660VD_3ST_DIG Signed-off-by: Mike Crash Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c5c3cfa5ec..4bff20f76b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -122,6 +122,7 @@ enum { /* ALC861-VD models */ enum { ALC660VD_3ST, + ALC660VD_3ST_DIG, ALC861VD_3ST, ALC861VD_3ST_DIG, ALC861VD_6ST_DIG, @@ -9987,6 +9988,7 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re */ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", + [ALC660VD_3ST_DIG]= "3stack-660-digout", [ALC861VD_3ST] = "3stack", [ALC861VD_3ST_DIG] = "3stack-digout", [ALC861VD_6ST_DIG] = "6stack-digout", @@ -9998,7 +10000,7 @@ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), @@ -10023,6 +10025,19 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, }, + [ALC660VD_3ST_DIG] = { + .mixers = { alc861vd_3st_mixer }, + .init_verbs = { alc861vd_volume_init_verbs, + alc861vd_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), + .dac_nids = alc660vd_dac_nids, + .dig_out_nid = ALC861VD_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .adc_nids = alc861vd_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_capture_source, + }, [ALC861VD_3ST] = { .mixers = { alc861vd_3st_mixer }, .init_verbs = { alc861vd_volume_init_verbs, -- cgit v1.2.3 From f8f794ab27dd3c6d9965ec5434775ead23d3d7a2 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Mon, 25 Jun 2007 12:14:25 +0200 Subject: [ALSA] HDA - Change default configuration for Fujitsu Si3515 This patch changes the default configuration for the Fujitsu Siemens Si3515 Laptop to the correct model=laptop. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 90ebc2c4988..0ffcf2001d2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -802,7 +802,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), {} }; -- cgit v1.2.3 From 076c0e4fd80a43f64710fe9aabdc4c463b7f8621 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 26 Jun 2007 11:43:52 +0200 Subject: [ALSA] fix ensoniq section mismatch Fix ensoniq driver sections when HOTPLUG=n, but code is used during resume, so it cannot be discarded: WARNING: sound/pci/snd-ens1371.o(.text+0x5c0): Section mismatch: reference to .init.text: (between 'snd_ensoniq_chip_init' and 'snd_ensoniq_free') Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ens1370.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 7c403965153..21cb4268a59 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1607,8 +1607,8 @@ struct es1371_quirk { unsigned char rev; /* revision */ }; -static int __devinit es1371_quirk_lookup(struct ensoniq *ensoniq, - struct es1371_quirk *list) +static int es1371_quirk_lookup(struct ensoniq *ensoniq, + struct es1371_quirk *list) { while (list->vid != (unsigned short)PCI_ANY_ID) { if (ensoniq->pci->vendor == list->vid && -- cgit v1.2.3 From d10ad1a26da03fe089c02846a95fa5938c3ca3ce Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Mon, 2 Jul 2007 16:03:12 +0200 Subject: [ALSA] sound/pci/hda/patch_realtek.c: remove dead code This patch removes some dead code spotted by the Coverity checker. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4bff20f76b0..5367d1377e1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11182,8 +11182,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - if (err < 0) - return err; spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; -- cgit v1.2.3 From df802952b723cdb08f2f8a3cba396c357290850b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jul 2007 19:18:00 +0200 Subject: [ALSA] hda-codec - Fix the error in DAC assignment of STAC codec The STAC codec support code has the automatic recognition of additional side-output pins, and this may result in an error when the assigned pin has no corresponding DAC connection. The patch fixes the problem -- it corrects the line_outs again without such additional pins. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 270539a2bbc..e89180c6aaa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1228,7 +1228,7 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) * and 9202/925x. For those, dac_nids[] must be hard-coded. */ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) + struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; int i, j, conn_len = 0; @@ -1253,6 +1253,13 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, } if (j == conn_len) { + if (spec->multiout.num_dacs > 0) { + /* we have already working output pins, + * so let's drop the broken ones again + */ + cfg->line_outs = spec->multiout.num_dacs; + break; + } /* error out, no available DAC found */ snd_printk(KERN_ERR "%s: No available DAC for pin 0x%x\n", -- cgit v1.2.3 From 8481da5a8d009d9bbac3d1483b579940e6dd9d59 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Tue, 3 Jul 2007 20:05:19 +0200 Subject: [ALSA] hda-codec - Add support for HP Spartan Added the model for HP Spartan with Conexant codec chip. From: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0ffcf2001d2..4d8e8af5c81 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -801,6 +801,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), -- cgit v1.2.3 From 8f41b56645f03e6f50407211d6f0f0627ae95e4e Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Tue, 3 Jul 2007 20:02:49 +0200 Subject: [ALSA] hda-codec - Add support for HP Nettle Added the model for HP Nettle mobo with ALC883. From: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5367d1377e1..a4876a31311 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6410,6 +6410,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; -- cgit v1.2.3 From 4210861964145617cf27b2b9c45860bff3e8290c Mon Sep 17 00:00:00 2001 From: Paul Vojta Date: Wed, 4 Jul 2007 10:35:29 +0200 Subject: [ALSA] nm256 - Add mention of opl3sa2 to a diagnostic message Adds mention of opl3sa2 driver to a diagnostic message for NeoMagic nm256 driver. Signed-off-by: Paul Vojta Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/nm256/nm256.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 03b3a4792f7..c7621bd770a 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1533,7 +1533,8 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, printk(KERN_ERR " force the driver to load by " "passing in the module parameter\n"); printk(KERN_ERR " force_ac97=1\n"); - printk(KERN_ERR " or try sb16 or cs423x drivers instead.\n"); + printk(KERN_ERR " or try sb16, opl3sa2, or " + "cs423x drivers instead.\n"); err = -ENXIO; goto __error; } -- cgit v1.2.3 From 0a8c5da3c066a1e92a66a493e6fe04d5ce7a206d Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Wed, 4 Jul 2007 15:17:38 +0200 Subject: [ALSA] hda-codec - Fix 5.1 output in LG LW20 Implemented LG LW20 5.1 support based on ALC880 threestack modes. It was previously set to use one DAC and 2 jack modes, giving 2-channel output only. Analog CD playing support is still missing. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 32 ++++++++++++++++++++++++++------ 1 file changed, 26 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a4876a31311..8deca0f7d7c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1900,17 +1900,38 @@ static struct hda_input_mux alc880_lg_lw_capture_source = { }, }; +#define alc880_lg_lw_modes alc880_threestack_modes + static struct snd_kcontrol_new alc880_lg_lw_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, 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("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, { } /* end */ }; static struct hda_verb alc880_lg_lw_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ + {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ + /* set capture source to mic-in */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1920,7 +1941,6 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* HP-out */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* mic-in to input */ @@ -2864,11 +2884,11 @@ static struct alc_config_preset alc880_presets[] = { .mixers = { alc880_lg_lw_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_lg_lw_init_verbs }, - .num_dacs = 1, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), .dac_nids = alc880_dac_nids, .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, + .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, -- cgit v1.2.3 From 83c34218a867085f7108438408c8323b46dbfc4a Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 5 Jul 2007 11:43:05 +0200 Subject: [ALSA] hda-codec - Some fixes for Realtek codec supports - Added BenQ T31-16 AMD Turion64x2 (ALC262) support - Added models for ASUS machines with ALC861 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8deca0f7d7c..2d35b4cfb2b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -94,6 +94,7 @@ enum { ALC262_HP_BPC_D7000_WF, ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, + ALC262_BENQ_T31, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -6888,7 +6889,16 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { { } /* end */ }; - +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_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), + HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { } /* end */ +}; #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -7220,6 +7230,15 @@ static struct hda_verb alc262_EAPD_verbs[] = { {} }; +static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3050}, + {} +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -7641,6 +7660,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), + SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), @@ -7745,6 +7765,17 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + }, + [ALC262_BENQ_T31] = { + .mixers = { alc262_benq_t31_mixer }, + .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_hippo_unsol_event, }, }; @@ -9311,7 +9342,9 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), + SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), + SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), -- cgit v1.2.3 From 067b5a844c7b9703b0bdcd8cf003a621734c3bae Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 5 Jul 2007 12:35:15 +0200 Subject: [ALSA] hda-codec - Fix ALC662 auto mode Fixed ALC662 auto mode, wrong NIDs were chosen. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d35b4cfb2b..acaf8c39eea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11013,7 +11013,7 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, for (i = 0; i < cfg->line_outs; i++) { if (!spec->multiout.dac_nids[i]) continue; - nid = alc880_idx_to_dac(i); + nid = alc880_idx_to_mixer(i); if (i == 2) { /* Center/LFE */ err = add_control(spec, ALC_CTL_WIDGET_VOL, -- cgit v1.2.3 From 1624cb9a35d4276c8ea0d939d580a6945ac72ac1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Jul 2007 13:10:51 +0200 Subject: [ALSA] hda-codec - Fix the recording selection on VAIO laptop Fix the mismatch of init verb and capture mux with VAIO STAC9872. Also, changed the item names to be more intuitive. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e89180c6aaa..f20b4faf6cc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2332,8 +2332,8 @@ static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { /* { "HP", 0x0 }, */ - { "Line", 0x1 }, - { "Mic", 0x2 }, + { "Mic Jack", 0x1 }, + { "Internal Mic", 0x2 }, { "PCM", 0x3 }, } }; @@ -2344,7 +2344,7 @@ static struct hda_verb vaio_init[] = { {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ @@ -2360,7 +2360,7 @@ static struct hda_verb vaio_ar_init[] = { {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ /* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ /* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ -- cgit v1.2.3 From cd1e3b40429c172b29c04a9c777be2ca1ac012e0 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 6 Jul 2007 12:10:01 +0200 Subject: [ALSA] hda-codec - Fix HP Nettle 5.1 output Fixed HP Nettle 6ch/8ch channel routing based on ALC883 sixstack and swapping rear and center/LFE in verbs and mixer settings. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 65 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 64 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index acaf8c39eea..33c6fafd358 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -172,6 +172,7 @@ enum { ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, + ALC888_HP_NETTLE, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6024,6 +6025,46 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + 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_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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6155,6 +6196,14 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_hp_nettle_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ + { } +}; + /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { @@ -6397,6 +6446,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", + [ALC888_HP_NETTLE] = "hp-nettle", [ALC883_AUTO] = "auto", }; @@ -6431,7 +6481,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6616,6 +6666,19 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_lenovo_ms7195_unsol_event, .init_hook = alc888_lenovo_ms7195_front_automute, }, + [ALC888_HP_NETTLE] = { + .mixers = { alc888_hp_nettle_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_hp_nettle_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + }, }; -- cgit v1.2.3 From 8341de60c76c400eaa453c91810ba6995d2fdb57 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 6 Jul 2007 12:10:45 +0200 Subject: [ALSA] hda-codec - Add HP Lucknow 5.1 support Added HP Lucknow (ALC888) 5.1 surround support based on ALC883 6-channel threestack with changes in channel routing. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 81 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 81 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33c6fafd358..993b503a053 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -173,6 +173,7 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC888_HP_NETTLE, + ALC888_HP_LUCKNOW, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6065,6 +6066,44 @@ static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_hp_lucknow_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6204,6 +6243,34 @@ static struct hda_verb alc888_hp_nettle_verbs[] = { { } }; +static struct hda_verb alc888_hp_lucknow_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + { } +}; + +static struct hda_verb alc888_hp_lucknow_2ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } +}; + +static struct hda_verb alc888_hp_lucknow_6ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } +}; + +static struct hda_channel_mode alc888_hp_lucknow_modes[2] = { + { 2, alc888_hp_lucknow_2ch_init }, + { 6, alc888_hp_lucknow_6ch_init }, +}; + /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { @@ -6447,6 +6514,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC888_HP_NETTLE] = "hp-nettle", + [ALC888_HP_LUCKNOW] = "hp-lucknow", [ALC883_AUTO] = "auto", }; @@ -6482,6 +6550,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), + SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_HP_LUCKNOW), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6679,6 +6748,18 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC888_HP_LUCKNOW] = { + .mixers = { alc888_hp_lucknow_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_hp_lucknow_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc888_hp_lucknow_modes), + .channel_mode = alc888_hp_lucknow_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + }, }; -- cgit v1.2.3 From 0f40502e0b5300a37c4cfc7bf93f84c2702dbbe9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 12:24:11 +0200 Subject: [ALSA] hda-codec - Add missing model parameter strings for ALC262 Added the missing model option values for ALC262. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 993b503a053..2f074ba5748 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7778,7 +7778,8 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_BENQ_ED8] = "benq", - [ALC262_BENQ_ED8] = "sony-assamd", + [ALC262_BENQ_T31] = "benq-t31", + [ALC262_SONY_ASSAMD] = "sony-assamd", [ALC262_AUTO] = "auto", }; -- cgit v1.2.3 From a4eed138add1018846d17e813560b0c7c0ae8e01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 18:17:04 +0200 Subject: [ALSA] hda-codec - Fix number of pin widgets with STAC codecs Use ARRAY_SIZE() to calculate the number of pin widgets for each codec chip. This also fixes a typo in STAC9205 code that gives a wrong number (14 instead of 12), which resulted in fatal communication errors. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f20b4faf6cc..0dccba231e8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1958,7 +1958,7 @@ static int patch_stac9200(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 8; + spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, @@ -2008,7 +2008,7 @@ static int patch_stac925x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 8; + spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS, stac925x_models, @@ -2080,7 +2080,7 @@ static int patch_stac922x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 10; + spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, @@ -2181,7 +2181,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 14; + spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, stac927x_models, @@ -2266,7 +2266,7 @@ static int patch_stac9205(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 14; + spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS, stac9205_models, -- cgit v1.2.3 From 43bbb6ccc67775b432c31d5fd2e1050329ffdcbf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Jul 2007 20:22:05 +0200 Subject: [ALSA] hda-intel - Show the last command in warning messages Show the last issued command in warning messages when any communication error with CORB/RIRB occurs. Also, a bit code clean-up for composing the command value in *_send_cmd(). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 45 +++++++++++++++++++++------------------------ 1 file changed, 21 insertions(+), 24 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c78ff901a57..92bc8b3fa2a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -341,6 +341,9 @@ struct azx { unsigned int single_cmd :1; unsigned int polling_mode :1; unsigned int msi :1; + + /* for debugging */ + unsigned int last_cmd; /* last issued command (to sync) */ }; /* driver types */ @@ -466,18 +469,10 @@ static void azx_free_cmd_io(struct azx *chip) } /* send a command */ -static int azx_corb_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int para) +static int azx_corb_send_cmd(struct hda_codec *codec, u32 val) { struct azx *chip = codec->bus->private_data; unsigned int wp; - u32 val; - - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; /* add command to corb */ wp = azx_readb(chip, CORBWP); @@ -543,7 +538,7 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) if (chip->msi) { snd_printk(KERN_WARNING "hda_intel: No response from codec, " - "disabling MSI...\n"); + "disabling MSI: last cmd=0x%08x\n", chip->last_cmd); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -555,13 +550,15 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) if (!chip->polling_mode) { snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, " - "switching to polling mode...\n"); + "switching to polling mode: last cmd=0x%08x\n", + chip->last_cmd); chip->polling_mode = 1; goto again; } snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " - "switching to single_cmd mode...\n"); + "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 */ @@ -581,20 +578,11 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) */ /* send a command */ -static int azx_single_send_cmd(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, - unsigned int para) +static int azx_single_send_cmd(struct hda_codec *codec, u32 val) { struct azx *chip = codec->bus->private_data; - u32 val; int timeout = 50; - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; - while (timeout--) { /* check ICB busy bit */ if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) { @@ -639,10 +627,19 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, unsigned int para) { struct azx *chip = codec->bus->private_data; + u32 val; + + val = (u32)(codec->addr & 0x0f) << 28; + val |= (u32)direct << 27; + val |= (u32)nid << 20; + val |= verb << 8; + val |= para; + chip->last_cmd = val; + if (chip->single_cmd) - return azx_single_send_cmd(codec, nid, direct, verb, para); + return azx_single_send_cmd(codec, val); else - return azx_corb_send_cmd(codec, nid, direct, verb, para); + return azx_corb_send_cmd(codec, val); } /* get a response */ -- cgit v1.2.3 From 341d4eb0cffdcd5fbdbd696dc26671f91e7c418c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Jul 2007 17:53:18 +0200 Subject: [ALSA] hda-codec - Fix Toshiba A100 with ALC861 One of the newly added model entries causes a regression on Toshiba A100. It works fine with model=auto. So, let's deselect the entry again. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f074ba5748..80427668532 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9491,7 +9491,10 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), + /* FIXME: the entry below breaks Toshiba A100 (model=auto works!) + * Any other models that need this preset? + */ + /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */ SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), -- cgit v1.2.3 From f95474ec05106a2af46220fbf8624adc17fcb3e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jul 2007 00:47:43 +0200 Subject: [ALSA] hda-codec - Fix default pin config of Abit AW9D-MAX Fix the default pin configuration of Abit AW9D-MAX. The board has a broken BIOS that doesn't set the correct pin default configs, which screws up the auto-configuration of snd-hda-intel driver. The patch enables the override of default pin config values. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 57 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80427668532..d7eb22f5ca3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -723,6 +723,38 @@ static void alc_subsystem_id(struct hda_codec *codec, } } +/* + * Fix-up pin default configurations + */ + +struct alc_pincfg { + hda_nid_t nid; + u32 val; +}; + +static void alc_fix_pincfg(struct hda_codec *codec, + const struct snd_pci_quirk *quirk, + const struct alc_pincfg **pinfix) +{ + const struct alc_pincfg *cfg; + + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (!quirk) + return; + + cfg = pinfix[quirk->value]; + for (; cfg->nid; cfg++) { + int i; + u32 val = cfg->val; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, cfg->nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + val & 0xff); + val >>= 8; + } + } +} + /* * ALC880 3-stack model * @@ -5409,6 +5441,29 @@ static struct alc_config_preset alc882_presets[] = { }; +/* + * Pin config fixes + */ +enum { + PINFIX_ABIT_AW9D_MAX +}; + +static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { + { 0x15, 0x01080104 }, /* side */ + { 0x16, 0x01011012 }, /* rear */ + { 0x17, 0x01016011 }, /* clfe */ + { } +}; + +static const struct alc_pincfg *alc882_pin_fixes[] = { + [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +}; + +static struct snd_pci_quirk alc882_pinfix_tbl[] = { + SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), + {} +}; + /* * BIOS auto configuration */ @@ -5532,6 +5587,8 @@ static int patch_alc882(struct hda_codec *codec) } } + alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ err = alc882_parse_auto_config(codec); -- cgit v1.2.3 From 2dcd522f89d17936d08665c22ab0ee415b8c5d56 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Tue, 10 Jul 2007 17:04:57 +0200 Subject: [ALSA] HDA-Intel - Add support for MSI K9AGM2-FIH motherboard This patch adds support for the MSI K9AGM2-FIH on-board audio. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_atihdmi.c | 1 + sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index f8eb4c90f80..72d3ab9751a 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -172,6 +172,7 @@ static int patch_atihdmi(struct hda_codec *codec) */ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, + { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R600 HDMI", .patch = patch_atihdmi }, {} /* terminator */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d7eb22f5ca3..b145667de17 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6586,6 +6586,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), -- cgit v1.2.3 From be38114a49853e441e694d690e631937de5fc524 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Wed, 11 Jul 2007 10:31:08 +0200 Subject: [ALSA] hda-codec - Add VIA HDA to si3054 Add VIA HDA identification to use the HDA-based Motorola modem of the Clevo m540 laptop. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_si3054.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 43f537ef40b..5dd9f61b7cb 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -304,6 +304,8 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 }, + /* VIA HDA on Clevo m540 */ + { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 }, /* Asus A8J Modem (SM56) */ { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, {} -- cgit v1.2.3 From ef2ec0dd65bc542f9ab45ea2fac6920cb1afa13b Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Thu, 12 Jul 2007 11:27:37 +0200 Subject: [ALSA] hda-codec - Add LG LW20 si3054 modem id Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_si3054.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 5dd9f61b7cb..6d2ecc38905 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -308,6 +308,8 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 }, /* Asus A8J Modem (SM56) */ { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, + /* LG LW20 modem */ + { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 }, {} }; -- cgit v1.2.3 From 1b290a510c12be91b6a2f7d6a07ef61b8ee9f06a Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 12 Jul 2007 15:17:34 +0200 Subject: [ALSA] hda: stac9202 mixer fix Added support for line out controls for STAC9202 cards, and fixed issue where master mixer control was being created twice for headphone and speaker outs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0dccba231e8..be53e9685bc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -334,8 +334,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -1401,7 +1399,15 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, continue; add_spec_dacs(spec, nid); } - + for (i = 0; i < cfg->line_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); + } for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2", -- cgit v1.2.3 From e4f41da9e5f03e67ebe6067aad40c6a79443c3ae Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 13 Jul 2007 11:51:06 +0200 Subject: [ALSA] hda-codec - Add LG LW20 line-in capture source Add line-in to the list of LG LW20 capture sources. Also fix the LG LW pin assignment list comment. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b145667de17..e8782c27a45 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1921,17 +1921,18 @@ static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) * Pin assignment: * Speaker-out: 0x14 * Mic-In: 0x18 - * Built-in Mic-In: 0x19 (?) - * HP-Out: 0x1b + * Built-in Mic-In: 0x19 + * Line-In: 0x1b + * HP-Out: 0x1a * SPDIF-Out: 0x1e */ -/* seems analog CD is not working */ static struct hda_input_mux alc880_lg_lw_capture_source = { - .num_items = 2, + .num_items = 3, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, + { "Line In", 0x2 }, }, }; -- cgit v1.2.3 From e87f97a3e842a816f30f087d5bfac68ef2afaef2 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 13 Jul 2007 11:51:32 +0200 Subject: [ALSA] hda-codec - Add quirk for HP Samba This patch fixes 5.1 surround output for the HP Samba using the same quirk as the HP Lucknow. If HP machines are uniform in their wiring, we should rename ALC888_HP_NETTLE to ALC888_6ST_HP and ALC888_HP_LUCKNOW to ALC888_3ST_HP for generic HP 6stack and 3stack configurations. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e8782c27a45..fa259a98ea7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6610,6 +6610,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_HP_LUCKNOW), + SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_HP_LUCKNOW), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; -- cgit v1.2.3 From 4723c022e1960abb5608b37ef61443da7d6296bb Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Fri, 13 Jul 2007 14:36:19 +0200 Subject: [ALSA] hda-codec - Rename HP model-specific quirks Rename ALC888_HP_NETTLE and ALC888_HP_LUCKNOW models to the more generic names ALC888_6ST_HP and ALC888_3ST_HP since HP seems to be consistent in the wiring of their 3stack and 6stack ALC888-based systems. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 48 +++++++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 24 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa259a98ea7..d427c32301f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -172,8 +172,8 @@ enum { ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, - ALC888_HP_NETTLE, - ALC888_HP_LUCKNOW, + ALC888_6ST_HP, + ALC888_3ST_HP, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6084,7 +6084,7 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { +static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -6124,7 +6124,7 @@ static struct snd_kcontrol_new alc888_hp_nettle_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_hp_lucknow_mixer[] = { +static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -6293,7 +6293,7 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_hp_nettle_verbs[] = { +static struct hda_verb alc888_6st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ @@ -6301,14 +6301,14 @@ static struct hda_verb alc888_hp_nettle_verbs[] = { { } }; -static struct hda_verb alc888_hp_lucknow_verbs[] = { +static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; -static struct hda_verb alc888_hp_lucknow_2ch_init[] = { +static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -6316,7 +6316,7 @@ static struct hda_verb alc888_hp_lucknow_2ch_init[] = { { } }; -static struct hda_verb alc888_hp_lucknow_6ch_init[] = { +static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -6324,9 +6324,9 @@ static struct hda_verb alc888_hp_lucknow_6ch_init[] = { { } }; -static struct hda_channel_mode alc888_hp_lucknow_modes[2] = { - { 2, alc888_hp_lucknow_2ch_init }, - { 6, alc888_hp_lucknow_6ch_init }, +static struct hda_channel_mode alc888_3st_hp_modes[2] = { + { 2, alc888_3st_hp_2ch_init }, + { 6, alc888_3st_hp_6ch_init }, }; /* toggle front-jack and RCA according to the hp-jack state */ @@ -6571,8 +6571,8 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", - [ALC888_HP_NETTLE] = "hp-nettle", - [ALC888_HP_LUCKNOW] = "hp-lucknow", + [ALC888_6ST_HP] = "6stack-hp", + [ALC888_3ST_HP] = "3stack-hp", [ALC883_AUTO] = "auto", }; @@ -6608,9 +6608,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_HP_NETTLE), - SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_HP_LUCKNOW), - SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_HP_LUCKNOW), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6795,9 +6795,9 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_lenovo_ms7195_unsol_event, .init_hook = alc888_lenovo_ms7195_front_automute, }, - [ALC888_HP_NETTLE] = { - .mixers = { alc888_hp_nettle_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_hp_nettle_verbs }, + [ALC888_6ST_HP] = { + .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -6808,15 +6808,15 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, - [ALC888_HP_LUCKNOW] = { - .mixers = { alc888_hp_lucknow_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_hp_lucknow_verbs }, + [ALC888_3ST_HP] = { + .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), .adc_nids = alc883_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc888_hp_lucknow_modes), - .channel_mode = alc888_hp_lucknow_modes, + .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes), + .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, -- cgit v1.2.3 From ef64adbbba1b215e8b523b50ef9d568b4bf2e1e3 Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Sat, 14 Jul 2007 00:26:16 +0200 Subject: [ALSA] hda-codec - Add quirks for HP dx2200/dx2250 HP dx2200 and dx2250 use Micro-Star International (MSI) motherboards (models MS-7254 and MS-7297 respectively) with an ALC862 codec in threestack configuration. Adding this quirk allows correct 5.1 sound output in these systems. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d427c32301f..28fa99da191 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9560,6 +9560,8 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), {} }; -- cgit v1.2.3 From 87d483630a4bd5af2beb2155c7ae8c408729a1a6 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 17 Jul 2007 11:52:24 +0200 Subject: [ALSA] hda: Enable SPDIF in/out on some stac9205 boards Added support for some boards with SPDIF in/out, and cleaned up the GPIO enable function. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 95 +++++++++++++++++++++++++++--------------- 1 file changed, 61 insertions(+), 34 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index be53e9685bc..9b1d433178b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -44,6 +44,7 @@ enum { enum { STAC_9205_REF, + STAC_M43xx, STAC_9205_MODELS }; @@ -218,7 +219,6 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x16, 0x17, 0x18, 0x21, 0x22, - }; static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, @@ -773,7 +773,8 @@ static unsigned int ref9205_pin_configs[12] = { }; static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { - ref9205_pin_configs, + [STAC_REF] = ref9205_pin_configs, + [STAC_M43xx] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -784,6 +785,10 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8, + "Dell Precision", STAC_M43xx), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff, + "Dell Precision", STAC_M43xx), {} /* terminator */ }; @@ -813,48 +818,56 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) return 0; } +static void stac92xx_set_config_reg(struct hda_codec *codec, + hda_nid_t pin_nid, unsigned int pin_config) +{ + int i; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + i = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, + 0x00); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", + pin_nid, i); +} + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; - unsigned int pin_cfg; - if (! spec->pin_nids || ! spec->pin_configs) - return; + if (!spec->pin_configs) + return; - for (i = 0; i < spec->num_pins; i++) { - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - spec->pin_configs[i] & 0x000000ff); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (spec->pin_configs[i] & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (spec->pin_configs[i] & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - spec->pin_configs[i] >> 24); - pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); - } + for (i = 0; i < spec->num_pins; i++) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); } -static void stac92xx_enable_eapd(struct hda_codec *codec) +static void stac92xx_enable_gpio_mask(struct hda_codec *codec, + int gpio_mask, int gpio_data) { - /* Configure GPIO0 as output */ + /* Configure GPIOx as output */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ + AC_VERB_SET_GPIO_DIRECTION, gpio_mask); + /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ + /* Assert GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ + AC_VERB_SET_GPIO_DATA, gpio_data); + /* Enable GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + AC_VERB_SET_GPIO_MASK, gpio_mask); } /* @@ -2233,7 +2246,8 @@ static int patch_stac927x(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); + /* GPIO0 High = Enable EAPD */ + stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { @@ -2265,7 +2279,7 @@ static int patch_stac927x(struct hda_codec *codec) static int patch_stac9205(struct hda_codec *codec) { struct sigmatel_spec *spec; - int err; + int err, gpio_mask, gpio_data; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2302,8 +2316,21 @@ static int patch_stac9205(struct hda_codec *codec) spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); + + if (spec->board_config == STAC_M43xx) { + /* Enable SPDIF in/out */ + stac92xx_set_config_reg(codec, 0x1f, 0x01441030); + stac92xx_set_config_reg(codec, 0x20, 0x1c410030); + + gpio_mask = 0x00000007; /* GPIO0-2 */ + /* GPIO0 High = EAPD, GPIO1 Low = DRM, + * GPIO2 High = Headphone Mute + */ + gpio_data = 0x00000005; + } else + gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { if (spec->board_config < 0) { -- cgit v1.2.3 From 89f157d9e6bf08b65f93dcb1dca1de037079885f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jul 2007 11:23:21 +0200 Subject: [ALSA] cs46xx - Fix PM resume Fixed PM resume of cs46xx devices. It now restores properly the DSP image and kick-off the DSP. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/cs46xx/cs46xx_lib.c | 77 +++++++++++++++---- sound/pci/cs46xx/cs46xx_lib.h | 3 + sound/pci/cs46xx/dsp_spos.c | 170 +++++++++++++++++++++++++++++++----------- 3 files changed, 191 insertions(+), 59 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index bef1f6d1859..71d7aab9d86 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2897,6 +2897,10 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) } #endif +#ifdef CONFIG_PM + kfree(chip->saved_regs); +#endif + pci_disable_device(chip->pci); kfree(chip); return 0; @@ -3140,6 +3144,23 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) /* * start and load DSP */ + +static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) +{ + unsigned int tmp; + + snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); + + tmp = snd_cs46xx_peek(chip, BA1_PFIE); + tmp &= ~0x0000f03f; + snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ + + tmp = snd_cs46xx_peek(chip, BA1_CIE); + tmp &= ~0x0000003f; + tmp |= 0x00000001; + snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ +} + int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) { unsigned int tmp; @@ -3214,19 +3235,7 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) snd_cs46xx_proc_start(chip); - /* - * Enable interrupts on the part. - */ - snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); - - tmp = snd_cs46xx_peek(chip, BA1_PFIE); - tmp &= ~0x0000f03f; - snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ - - tmp = snd_cs46xx_peek(chip, BA1_CIE); - tmp &= ~0x0000003f; - tmp |= 0x00000001; - snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ + cs46xx_enable_stream_irqs(chip); #ifndef CONFIG_SND_CS46XX_NEW_DSP /* set the attenuation to 0dB */ @@ -3665,11 +3674,19 @@ static struct cs_card_type __devinitdata cards[] = { * APM support */ #ifdef CONFIG_PM +static unsigned int saved_regs[] = { + BA0_ACOSV, + BA0_ASER_FADDR, + BA0_ASER_MASTER, + BA1_PVOL, + BA1_CVOL, +}; + int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; @@ -3680,6 +3697,10 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* save some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + chip->saved_regs[i] = snd_cs46xx_peekBA0(chip, saved_regs[i]); + amp_saved = chip->amplifier; /* turn off amp */ chip->amplifier_ctrl(chip, -chip->amplifier); @@ -3698,7 +3719,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3716,6 +3737,16 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_cs46xx_chip_init(chip); + snd_cs46xx_reset(chip); +#ifdef CONFIG_SND_CS46XX_NEW_DSP + cs46xx_dsp_resume(chip); + /* restore some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]); +#else + snd_cs46xx_download_image(chip); +#endif + #if 0 snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, chip->ac97_general_purpose); @@ -3730,6 +3761,13 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* reset playback/capture */ + snd_cs46xx_set_play_sample_rate(chip, 8000); + snd_cs46xx_set_capture_sample_rate(chip, 8000); + snd_cs46xx_proc_start(chip); + + cs46xx_enable_stream_irqs(chip); + if (amp_saved) chip->amplifier_ctrl(chip, 1); /* turn amp on */ else @@ -3896,6 +3934,15 @@ int __devinit snd_cs46xx_create(struct snd_card *card, snd_cs46xx_proc_init(card, chip); +#ifdef CONFIG_PM + chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) * + ARRAY_SIZE(saved_regs), GFP_KERNEL); + if (!chip->saved_regs) { + snd_cs46xx_free(chip); + return -ENOMEM; + } +#endif + chip->active_ctrl(chip, -1); /* disable CLKRUN */ snd_card_set_dev(card, &pci->dev); diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index f75750c2bd2..20dcd72f06c 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -86,6 +86,9 @@ static inline unsigned int snd_cs46xx_peekBA0(struct snd_cs46xx *chip, unsigned struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip); void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip); int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module); +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip); +#endif struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symbol_type); #ifdef CONFIG_PROC_FS diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 336e77e2600..590b35d91df 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -306,13 +306,59 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) mutex_unlock(&chip->spos_mutex); } +static int dsp_load_parameter(struct snd_cs46xx *chip, + struct dsp_segment_desc *parameter) +{ + u32 doffset, dsize; + + if (!parameter) { + snd_printdd("dsp_spos: module got no parameter segment\n"); + return 0; + } + + doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); + dsize = parameter->size * 4; + + snd_printdd("dsp_spos: " + "downloading parameter data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { + snd_printk(KERN_ERR "dsp_spos: " + "failed to download parameter data to DSP\n"); + return -EINVAL; + } + return 0; +} + +static int dsp_load_sample(struct snd_cs46xx *chip, + struct dsp_segment_desc *sample) +{ + u32 doffset, dsize; + + if (!sample) { + snd_printdd("dsp_spos: module got no sample segment\n"); + return 0; + } + + doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); + dsize = sample->size * 4; + + snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + + if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { + snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); + return -EINVAL; + } + return 0; +} + int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM); - struct dsp_segment_desc * parameter = get_segment_desc (module,SEGTYPE_SP_PARAMETER); - struct dsp_segment_desc * sample = get_segment_desc (module,SEGTYPE_SP_SAMPLE); u32 doffset, dsize; + int err; if (ins->nmodules == DSP_MAX_MODULES - 1) { snd_printk(KERN_ERR "dsp_spos: to many modules loaded into DSP\n"); @@ -326,49 +372,20 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE); } - if (parameter == NULL) { - snd_printdd("dsp_spos: module got no parameter segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current parameter data may be overwriten!\n"); - } - - doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); - dsize = parameter->size * 4; - - snd_printdd("dsp_spos: downloading parameter data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to download parameter data to DSP\n"); - return -EINVAL; - } - } + err = dsp_load_parameter(chip, get_segment_desc(module, + SEGTYPE_SP_PARAMETER)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing sample area\n"); snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, DSP_SAMPLE_BYTE_SIZE); } - if (sample == NULL) { - snd_printdd("dsp_spos: module got no sample segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current sample data may be overwriten\n"); - } - - doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); - dsize = sample->size * 4; - - snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); - return -EINVAL; - } - } - + err = dsp_load_sample(chip, get_segment_desc(module, + SEGTYPE_SP_SAMPLE)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing code area\n"); @@ -986,7 +1003,10 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return NULL; } - strcpy(ins->tasks[ins->ntask].task_name,name); + if (name) + strcpy(ins->tasks[ins->ntask].task_name, name); + else + strcpy(ins->tasks[ins->ntask].task_name, "(NULL)"); ins->tasks[ins->ntask].address = dest; ins->tasks[ins->ntask].size = size; @@ -995,7 +1015,8 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) desc = (ins->tasks + ins->ntask); ins->ntask++; - add_symbol (chip,name,dest,SYMBOL_PARAMETER); + if (name) + add_symbol (chip,name,dest,SYMBOL_PARAMETER); return desc; } @@ -1006,6 +1027,7 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 desc = _map_scb (chip,name,dest); if (desc) { + desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); @@ -1023,6 +1045,7 @@ cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_da desc = _map_task_tree (chip,name,dest,size); if (desc) { + desc->data = task_data; _dsp_create_task_tree(chip,task_data,dest,size); } else { snd_printk(KERN_ERR "dsp_spos: failed to map TASK\n"); @@ -1320,8 +1343,10 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) 0x0000ffff }; - /* dirty hack ... */ - _dsp_create_task_tree (chip,(u32 *)&mix2_ostream_spb,WRITE_BACK_SPB,2); + if (!cs46xx_dsp_create_task_tree(chip, NULL, + (u32 *)&mix2_ostream_spb, + WRITE_BACK_SPB, 2)) + goto _fail_end; } /* input sample converter */ @@ -1622,7 +1647,6 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, return 0; } - static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; @@ -1894,3 +1918,61 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) return 0; } + +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip) +{ + struct dsp_spos_instance * ins = chip->dsp_spos_instance; + int i, err; + + /* clear parameter, sample and code areas */ + snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, + DSP_PARAMETER_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, + DSP_SAMPLE_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE); + + for (i = 0; i < ins->nmodules; i++) { + struct dsp_module_desc *module = &ins->modules[i]; + struct dsp_segment_desc *seg; + u32 doffset, dsize; + + seg = get_segment_desc(module, SEGTYPE_SP_PARAMETER); + err = dsp_load_parameter(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_SAMPLE); + err = dsp_load_sample(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_PROGRAM); + if (!seg) + continue; + + doffset = seg->offset * 4 + module->load_address * 4 + + DSP_CODE_BYTE_OFFSET; + dsize = seg->size * 4; + err = snd_cs46xx_download(chip, + ins->code.data + module->load_address, + doffset, dsize); + if (err < 0) + return err; + } + + for (i = 0; i < ins->ntask; i++) { + struct dsp_task_descriptor *t = &ins->tasks[i]; + _dsp_create_task_tree(chip, t->data, t->address, t->size); + } + + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + _dsp_create_scb(chip, s->data, s->address); + } + + return 0; +} +#endif -- cgit v1.2.3 From 3663d845e557989d09e856c1e9e708e80a976dd4 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 14 Jul 2007 02:18:26 +0100 Subject: [ALSA] snd-emu10k1: Initial support for E-Mu 1616 and 1616m. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/emu10k1_main.c | 45 ++++++++++++++++++++++++++++++++-------- 1 file changed, 36 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 72367411ee6..6135fc99fa5 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -51,9 +51,13 @@ #define HANA_FILENAME "emu/hana.fw" #define DOCK_FILENAME "emu/audio_dock.fw" +#define EMU1010B_FILENAME "emu/emu1010b.fw" +#define MICRO_DOCK_FILENAME "emu/micro_dock.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010B_FILENAME); +MODULE_FIRMWARE(MICRO_DOCK_FILENAME); /************************************************************************* @@ -660,10 +664,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return err; } snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); +#if 0 if (fw_entry->size != 0x133a4) { snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename); return -EINVAL; } +#endif /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -758,7 +764,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg1=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x1f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ @@ -766,19 +772,26 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg2=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x1f) == 0x15) { /* FPGA failed to return to programming mode */ return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); - if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); + return err; + } } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - if (reg != 0x55) { + if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); return -ENODEV; @@ -995,16 +1008,23 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } } + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if (reg != 0x55) { + if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); return 0; @@ -1282,6 +1302,13 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spi_dac = 1, .i2c_adc = 1, .spk71 = 1} , + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1 , + .emu1010 = 2} , {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", .id = "Audigy2", -- cgit v1.2.3 From d9e8a552d486eff3281e58754d126500782dcd74 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 14 Jul 2007 10:24:49 +0100 Subject: [ALSA] snd-emu10k1:Enable E-Mu 1616m notebook firmware loading. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/emu10k1_main.c | 36 ++++++++++++++++++++++++------------ 1 file changed, 24 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6135fc99fa5..404ae1be0a4 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -53,11 +53,13 @@ #define DOCK_FILENAME "emu/audio_dock.fw" #define EMU1010B_FILENAME "emu/emu1010b.fw" #define MICRO_DOCK_FILENAME "emu/micro_dock.fw" +#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); MODULE_FIRMWARE(EMU1010B_FILENAME); MODULE_FIRMWARE(MICRO_DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); /************************************************************************* @@ -764,7 +766,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg1=0x%x\n",reg); - if ((reg & 0x1f) == 0x15) { + if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ @@ -772,8 +774,9 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg2=0x%x\n",reg); - if ((reg & 0x1f) == 0x15) { + if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ + snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); @@ -787,11 +790,16 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); return err; } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME); + return err; + } } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - if ((reg & 0x1f) != 0x15) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); return -ENODEV; @@ -1016,6 +1024,10 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { return err; } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } } snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); @@ -1024,7 +1036,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if ((reg & 0x1f) != 0x15) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); return 0; @@ -1299,9 +1311,8 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0108_chip = 1, .ca_cardbus_chip = 1, - .spi_dac = 1, - .i2c_adc = 1, - .spk71 = 1} , + .spk71 = 1 , + .emu1010 = 3} , {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", .id = "EMU1010", @@ -1742,12 +1753,13 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->fx8010.extout_mask = extout_mask; emu->enable_ir = enable_ir; + if (emu->card_capabilities->ca_cardbus_chip) { + if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + goto error; + } if (emu->card_capabilities->ecard) { if ((err = snd_emu10k1_ecard_init(emu)) < 0) goto error; - } else if (emu->card_capabilities->ca_cardbus_chip) { - if ((err = snd_emu10k1_cardbus_init(emu)) < 0) - goto error; } else if (emu->card_capabilities->emu1010) { if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { snd_emu10k1_free(emu); @@ -1893,10 +1905,10 @@ void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu) void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) { + if (emu->card_capabilities->ca_cardbus_chip) + snd_emu10k1_cardbus_init(emu); if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); - else if (emu->card_capabilities->ca_cardbus_chip) - snd_emu10k1_cardbus_init(emu); else if (emu->card_capabilities->emu1010) snd_emu10k1_emu1010_init(emu); else -- cgit v1.2.3 From 8632649b1c991aab883a8538e493c33e362d077a Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Fri, 18 May 2007 20:35:50 +0100 Subject: [ALSA] snd-ca0106: Add support for X-Fi Extreme Audio. This card is just a normal SB Live 24bit, but under a different marketing name. Signed-off-by: James Courtier-Dutton Signed-off-by: Jaroslav Kysela --- sound/pci/ca0106/ca0106_main.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 9fd7b8a5b75..fcab8fb97e3 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -168,6 +168,25 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); #include "ca0106.h" static struct snd_ca0106_details ca0106_chip_details[] = { + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* + * CTRL:CA0111-WTLF + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + /* Tested: + * Playback on front, rear, center/lfe speakers + * Capture from Mic in. + * Not-Tested: + * Capture from Line in. + * Playback to digital out. + */ + { .serial = 0x10121102, + .name = "X-Fi Extreme Audio [SB0790]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ /* AudigyLS[SB0310] */ { .serial = 0x10021102, .name = "AudigyLS [SB0310]", -- cgit v1.2.3 From c5d9f1cdbf067b5500886c5171159e99657341ce Mon Sep 17 00:00:00 2001 From: Claudio Matsuoka Date: Thu, 19 Jul 2007 23:18:32 +0200 Subject: [ALSA] hda-codec - Add quirk for Asus P5LD2 Without the proper model setting, the Asus P5LD2 rear outputs remain completely silent unlike other systems where the front speakers usually work. This patch adds the P5LD2 to the quirk table. Signed-off-by: Claudio Matsuoka Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28fa99da191..5825d9958f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5348,6 +5348,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), {} -- cgit v1.2.3 From c54728d8aa59283ece86cb745e5085ece8b4eedb Mon Sep 17 00:00:00 2001 From: Nicola Fagnani Date: Thu, 19 Jul 2007 23:28:52 +0200 Subject: [ALSA] hda-intel: support for iMac 24'' released on 09/2006 Basic audio support for the iMac 24'' model released on 09/2006, including headphone jack detection with automatic speaker muting. This iMac uses the Realtek ALC885 codec, not a Sigmatel one as in other models. Functionality has been tested for internal speakers, headphone and microphone. Signed-off-by: Nicola Fagnani Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 74 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 73 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5825d9958f3..9a47eec5a27 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -153,6 +153,7 @@ enum { ALC882_TARGA, ALC882_ASUS_A7J, ALC885_MACPRO, + ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -5118,6 +5119,60 @@ static struct hda_verb alc882_macpro_init_verbs[] = { { } }; +/* iMac 24 mixer. */ +static struct snd_kcontrol_new alc885_imac24_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), + { } /* end */ +}; + +/* iMac 24 init verbs. */ +static struct hda_verb alc885_imac24_init_verbs[] = { + /* Internal speakers: output 0 (0x0c) */ + {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, 0x00}, + /* Internal speakers: output 0 (0x0c) */ + {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, 0x00}, + /* Headphone: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Front Mic: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +/* Toggle speaker-output according to the hp-jack state */ +static void alc885_imac24_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_update(codec, 0x18, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +/* 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); +} + static struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -5338,6 +5393,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC885_MACPRO] = "macpro", + [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", }; @@ -5410,6 +5466,19 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc885_imac24_unsol_event, + .init_hook = alc885_imac24_automute, + }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer, alc882_capture_mixer }, @@ -5582,6 +5651,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; + case 0x106b1000: /* iMac 24 */ + board_config = ALC885_IMAC24; + break; default: printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); @@ -5608,7 +5680,7 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (board_config == ALC885_MACPRO) { + if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) { alc882_gpio_mute(codec, 0, 0); alc882_gpio_mute(codec, 1, 0); } -- cgit v1.2.3 From aafc03398baefaac380b34a1ee6356e7eed8e663 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 May 2007 18:01:06 +0200 Subject: [ALSA] hda-codec - Fix input with STAC92xx The recent fix for STAC92xx surround outputs broke the input pin setting for shared line-in and mic jacks. This patch fixes the breakage. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9b1d433178b..3f25de72966 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2266,13 +2266,6 @@ static int patch_stac927x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; - /* Fix Mux capture level; max to 2 */ - snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, - (0 << AC_AMPCAP_OFFSET_SHIFT) | - (2 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (0 << AC_AMPCAP_MUTE_SHIFT)); - return 0; } -- cgit v1.2.3