From f6127efba1295b4668327b97014e678370028827 Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Tue, 15 Jul 2008 03:00:21 +0200 Subject: ALSA: add TriTech 28023 AC97 codec ID and Wolfson 9701 name. Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 07364c00768a..8c49a00a5e39 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -161,6 +161,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, +{ 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] @@ -169,7 +170,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, -{ 0x574d4c00, 0xffffffff, "WM9701A", NULL, NULL }, +{ 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, { 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, -- cgit v1.2.3-58-ga151 From 2b30a55d4d09254d6b25814bf6ac0b7843afdc99 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Tue, 15 Jul 2008 15:07:19 +0200 Subject: ALSA: Au1xpsc: psc not disabled when TX is idle TX idleness isn't tested, but RX twice. PSC is not disabled when TX is idle Signed-off-by: Roel Kluin Acked-by: Manuel Lauss Signed-off-by: Takashi Iwai --- sound/soc/au1x/psc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index ba4b5c199f21..9384702c7ebd 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -231,7 +231,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) /* if both TX and RX are idle, disable PSC */ stat = au_readl(I2S_STAT(pscdata)); - if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) { + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { au_writel(0, I2S_CFG(pscdata)); au_sync(); au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); -- cgit v1.2.3-58-ga151 From e785d3d8fb5fab744d67fac9966229bcdc52db45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jul 2008 16:28:43 +0200 Subject: ALSA: hda - Align BDL position adjustment parameter It seems NVidia and other hardwares require the alignment for period update timing. For satisfying this condition, align the position adjustment for delayed wake-up to the initial bdl_pos_adj value. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 16715a68ba5e..ef9f072b47fc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1047,9 +1047,13 @@ static int azx_setup_periods(struct azx *chip, pos_adj = bdl_pos_adj[chip->dev_index]; if (pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; + int pos_align = pos_adj; pos_adj = (pos_adj * runtime->rate + 47999) / 48000; if (!pos_adj) - pos_adj = 1; + pos_adj = pos_align; + else + pos_adj = ((pos_adj + pos_align - 1) / pos_align) * + pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { snd_printk(KERN_WARNING "Too big adjustment %d\n", -- cgit v1.2.3-58-ga151 From 810fd3f3f621fef9d1ac71b198d830fdeafbc1c3 Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Thu, 17 Jul 2008 09:22:29 +0200 Subject: ALSA: ens1370: SRC stands for Sample Rate Converter Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index fbf1124f7c79..33032a73ecc8 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -522,7 +522,7 @@ static unsigned int snd_es1371_wait_src_ready(struct ensoniq * ensoniq) return r; cond_resched(); } - snd_printk(KERN_ERR "wait source ready timeout 0x%lx [0x%x]\n", + snd_printk(KERN_ERR "wait src ready timeout 0x%lx [0x%x]\n", ES_REG(ensoniq, 1371_SMPRATE), r); return 0; } -- cgit v1.2.3-58-ga151 From 462dba28e1921f19319d11a44b7bb97e72da2a79 Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Thu, 17 Jul 2008 14:02:16 +0200 Subject: ALSA: ALSA: ens1370: communicate PCI device to AC97 communicate the ES137x PCI device to the AC97 code for its subsys_vendor/device values Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 33032a73ecc8..9bf95367c882 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1629,6 +1629,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ensoniq; ac97.private_free = snd_ensoniq_mixer_free_ac97; + ac97.pci = ensoniq->pci; ac97.scaps = AC97_SCAP_AUDIO; if ((err = snd_ac97_mixer(pbus, &ac97, &ensoniq->u.es1371.ac97)) < 0) return err; -- cgit v1.2.3-58-ga151 From 2927d6eeca0a5004d81fa5bedbdf3f2b1b842903 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Jul 2008 15:06:50 +0100 Subject: ALSA: ASoC: Refactor DAPM event handler The DAPM event callback code has many layers of indentation, taking it over 80 columns. Refactor the code to give less indentation in order to avoid checkpatch issues on further changes and exploding indentation. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 79 +++++++++++++++++++++++++++------------------------- 1 file changed, 41 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061c2a6b..17698ef58dfb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -586,45 +586,48 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) power_change = (w->power == power) ? 0: 1; w->power = power; + if (!power_change) + continue; + /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + dapm_update_bits(w); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } -- cgit v1.2.3-58-ga151 From 9dd8d812d3b4d208a769ca3cf23a7f9294632d0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Jul 2008 15:06:51 +0100 Subject: ALSA: ASoC: Factor PGA DAPM handling into main This allows pre and post event hooks to be provided for PGA widgets. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 17698ef58dfb..820347c9ae4b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -523,24 +523,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; } - /* programmable gain/attenuation */ - if (w->id == snd_soc_dapm_pga) { - int on; - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - w->power = on = (out != 0 && in != 0) ? 1 : 0; - - if (!on) - dapm_set_pga(w, on); /* lower volume to reduce pops */ - dapm_update_bits(w); - if (on) - dapm_set_pga(w, on); /* restore volume from zero */ - - continue; - } - /* pre and post event widgets */ if (w->id == snd_soc_dapm_pre) { if (!w->event) @@ -611,8 +593,16 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !power) + dapm_set_pga(w, power); + dapm_update_bits(w); + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && power) + dapm_set_pga(w, power); + /* power up post event */ if (power && w->event && (w->event_flags & SND_SOC_DAPM_POST_PMU)) { -- cgit v1.2.3-58-ga151 From 2522d7359301efadfb5744ebd3c623c3af4a7b30 Mon Sep 17 00:00:00 2001 From: Alexander Holler Date: Thu, 17 Jul 2008 23:36:15 +0200 Subject: ALSA: hda - Added support for Asus V1Sn Added the necessary ID for Asus V1Sn to patch_realtek.c to use ALC861VD_LENOVO on these laptops. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2807bc840d26..e5f8d3b699ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12994,6 +12994,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), -- cgit v1.2.3-58-ga151 From 82af6bc0986c5140efc875b2d91326031f0254ab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Jul 2008 23:37:20 +0200 Subject: ALSA: opti93x - Fix NULL dereference Probing non-existing device causes Oops with snd-opti93x driver due to NULL access in the destructor of the error path. Signed-off-by: Takashi Iwai Tested-by: Rene Herman Acked-by: Rene Herman Tested-by: Ingo Molnar Acked-by: Ingo Molnar --- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 41c047e665ec..d20abb286124 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -688,7 +688,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) if (chip) { #ifdef OPTi93X struct snd_cs4231 *codec = chip->codec; - if (codec->irq > 0) { + if (codec && codec->irq > 0) { disable_irq(codec->irq); free_irq(codec->irq, codec); } -- cgit v1.2.3-58-ga151 From 51f6baad264ca4bacdbf4fa25c676fa30d344bfa Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Fri, 18 Jul 2008 11:15:12 +0200 Subject: ALSA: opti9xx: no isapnp param for !CONFIG_PNP "isapnp" needs CONFIG_PNP to be useful. Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d20abb286124..0797ca441a37 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -68,7 +68,9 @@ MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)}," static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ //static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +#ifdef CONFIG_PNP static int isapnp = 1; /* Enable ISA PnP detection */ +#endif static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */ static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */ static long fm_port = SNDRV_DEFAULT_PORT1; /* 0x388 */ @@ -85,8 +87,10 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard."); //module_param(enable, bool, 0444); //MODULE_PARM_DESC(enable, "Enable opti9xx soundcard."); +#ifdef CONFIG_PNP module_param(isapnp, bool, 0444); MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif module_param(port, long, 0444); MODULE_PARM_DESC(port, "WSS port # for opti9xx driver."); module_param(mpu_port, long, 0444); -- cgit v1.2.3-58-ga151 From f53281e62a41ac176f050307c0d746a1183a68e8 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 18 Jul 2008 12:36:43 +0200 Subject: ALSA: hda - Add support of ASUS Eeepc P90* - Support ASUS_P900A = P703 - Support ASUS_P901 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 179 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 172 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5f8d3b699ec..f1cdce4c8a63 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -122,6 +122,8 @@ enum { /* ALC269 models */ enum { ALC269_BASIC, + ALC269_ASUS_EEEPC_P703, + ALC269_ASUS_EEEPC_P901, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -10946,7 +10948,23 @@ static int patch_alc268(struct hda_codec *codec) static hda_nid_t alc269_adc_nids[1] = { /* ADC1 */ - 0x07, + 0x08, +}; + +static struct hda_input_mux alc269_eeepc_dmic_capture_source = { + .num_items = 2, + .items = { + { "i-Mic", 0x5 }, + { "e-Mic", 0x0 }, + }, +}; + +static struct hda_input_mux alc269_eeepc_amic_capture_source = { + .num_items = 2, + .items = { + { "i-Mic", 0x1 }, + { "e-Mic", 0x0 }, + }, }; #define alc269_modes alc260_modes @@ -10968,10 +10986,27 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { { } /* end */ }; +/* bind volumes of both NID 0x0c and 0x0d */ +static struct hda_bind_ctls alc269_epc_bind_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc269_eeepc_mixer[] = { + HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol), + HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc269_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -10987,6 +11022,13 @@ static struct snd_kcontrol_new alc269_capture_mixer[] = { { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -10994,7 +11036,7 @@ static struct hda_verb alc269_init_verbs[] = { /* * Unmute ADC0 and set the default input 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)}, /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the * analog-loopback mixer widget @@ -11057,6 +11099,98 @@ static struct hda_verb alc269_init_verbs[] = { { } }; +static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc269_eeepc_amic_init_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc269_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned int bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); +} + +static void alc269_eeepc_dmic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 5); +} + +static void alc269_eeepc_amic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, + present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0)); + snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, + present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1)); +} + +/* unsolicited event for HP jack sensing */ +static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_speaker_automute(codec); + + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_eeepc_dmic_automute(codec); +} + +static void alc269_eeepc_dmic_inithook(struct hda_codec *codec) +{ + alc269_speaker_automute(codec); + alc269_eeepc_dmic_automute(codec); +} + +/* unsolicited event for HP jack sensing */ +static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_speaker_automute(codec); + + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_eeepc_amic_automute(codec); +} + +static void alc269_eeepc_amic_inithook(struct hda_codec *codec) +{ + alc269_speaker_automute(codec); + alc269_eeepc_amic_automute(codec); +} + /* add playback controls from the parsed DAC table */ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -11188,6 +11322,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + spec->mixers[spec->num_mixers] = alc269_capture_mixer; + spec->num_mixers++; + return 1; } @@ -11215,12 +11352,16 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { }; static struct snd_pci_quirk alc269_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", + ALC269_ASUS_EEEPC_P703), + SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", + ALC269_ASUS_EEEPC_P901), {} }; static struct alc_config_preset alc269_presets[] = { [ALC269_BASIC] = { - .mixers = { alc269_base_mixer }, + .mixers = { alc269_base_mixer, alc269_capture_mixer }, .init_verbs = { alc269_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, @@ -11229,6 +11370,32 @@ static struct alc_config_preset alc269_presets[] = { .channel_mode = alc269_modes, .input_mux = &alc269_capture_source, }, + [ALC269_ASUS_EEEPC_P703] = { + .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer }, + .init_verbs = { alc269_init_verbs, + alc269_eeepc_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_eeepc_amic_capture_source, + .unsol_event = alc269_eeepc_amic_unsol_event, + .init_hook = alc269_eeepc_amic_inithook, + }, + [ALC269_ASUS_EEEPC_P901] = { + .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer}, + .init_verbs = { alc269_init_verbs, + alc269_eeepc_dmic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_eeepc_dmic_capture_source, + .unsol_event = alc269_eeepc_dmic_unsol_event, + .init_hook = alc269_eeepc_dmic_inithook, + }, }; static int patch_alc269(struct hda_codec *codec) @@ -11282,8 +11449,6 @@ static int patch_alc269(struct hda_codec *codec) spec->adc_nids = alc269_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->mixers[spec->num_mixers] = alc269_capture_mixer; - spec->num_mixers++; codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) -- cgit v1.2.3-58-ga151 From 9432484110263e9418f380faf05fa9e2e7fb87a0 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Sun, 20 Jul 2008 17:36:20 +0200 Subject: ALSA: soc - wm9712 mono mixer this fixes typo in wm9712 codec which prevents it from registering all audio routes (and thus working correctly). Please consider applying. (Tested and works on palmtx, palmld and palmt5) Signed-off-by: Marek Vasut Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9712.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9fc8edd82225..1fb7f9a7aecd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -427,20 +427,20 @@ static const struct snd_soc_dapm_route audio_map[] = { {"HPOUTR", NULL, "Headphone PGA"}, {"Headphone PGA", NULL, "Right HP Mixer"}, - /* mono hp mixer */ - {"Mono HP Mixer", NULL, "Left HP Mixer"}, - {"Mono HP Mixer", NULL, "Right HP Mixer"}, + /* mono mixer */ + {"Mono Mixer", NULL, "Left HP Mixer"}, + {"Mono Mixer", NULL, "Right HP Mixer"}, /* Out3 Mux */ {"Out3 Mux", "Left", "Left HP Mixer"}, {"Out3 Mux", "Mono", "Phone Mixer"}, - {"Out3 Mux", "Left + Right", "Mono HP Mixer"}, + {"Out3 Mux", "Left + Right", "Mono Mixer"}, {"Out 3 PGA", NULL, "Out3 Mux"}, {"OUT3", NULL, "Out 3 PGA"}, /* speaker Mux */ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"}, - {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"}, + {"Speaker Mux", "Headphone Mix", "Mono Mixer"}, {"Speaker PGA", NULL, "Speaker Mux"}, {"LOUT2", NULL, "Speaker PGA"}, {"ROUT2", NULL, "Speaker PGA"}, -- cgit v1.2.3-58-ga151 From 6aa1e464453e398e4ab12558777fb10cff8a284d Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 22 Jul 2008 20:21:28 +0300 Subject: ALSA: sound/pci/azt3328.h: no variables for enums AZF_FREQUENCIES and AZF_GAME_CONFIGS were variables, and this doesn't seem to have been intended. Signed-off-by: Adrian Bunk Acked-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/azt3328.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 7e3e8942d073..974e05122f00 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -94,7 +94,7 @@ enum azf_freq_t { AZF_FREQ(48000), AZF_FREQ(66200), #undef AZF_FREQ -} AZF_FREQUENCIES; +}; /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ @@ -210,7 +210,7 @@ enum azf_freq_t { enum { AZF_GAME_LEGACY_IO_PORT = 0x200 -} AZF_GAME_CONFIGS; +}; #define IDX_GAME_LEGACY_COMPATIBLE 0x00 /* in some operation mode, writing anything to this port -- cgit v1.2.3-58-ga151 From 13c2108de4437771a77f775fe33e9a33c53a8a14 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 22 Jul 2008 20:21:32 +0300 Subject: ALSA: make snd_ac97_add_vmaster() static This patch makes the needlessly global snd_ac97_add_vmaster() static. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 0746e9ccc20b..f4fbc795ee81 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3381,8 +3381,8 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, } /* create a virtual master control and add slaves */ -int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, - const unsigned int *tlv, const char **slaves) +static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) { struct snd_kcontrol *kctl; const char **s; -- cgit v1.2.3-58-ga151 From fe7e873f52f17ad9b8ee9e2c70acaddcae22443b Mon Sep 17 00:00:00 2001 From: Travis Place Date: Sun, 27 Jul 2008 10:13:26 +0200 Subject: ALSA: hda - Add automatic model setting for the Acer Aspire 5920G laptop Make the Acer Aspire 5920G (1025:0121) select ALC883_ACER_ASPIRE by default. Signed-off-by: Travis Place Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1cdce4c8a63..add4e87e0b20 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7907,6 +7907,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), -- cgit v1.2.3-58-ga151 From b15ebe2616289da258f85b3ff142fca237ef9f59 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 23 Jul 2008 07:48:49 +0200 Subject: ALSA: cs4232: fix crash during chip PNP detection The acard->wss pointer is uninitialized in this function which leads to crash during chip PNP detection. Signed-off-by: Krzysztof Helt Acked-by: Rene Herman Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index dbe63db4bfd6..4d4b8ddc26ba 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -325,6 +325,7 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, struct pnp_dev *pdev) { + acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; cport[dev] = -1; -- cgit v1.2.3-58-ga151 From 536319afd1f25383009c0c88f6fb00104f49c178 Mon Sep 17 00:00:00 2001 From: Nicolas Boichat Date: Mon, 21 Jul 2008 22:18:01 +0800 Subject: ALSA: Allow to force model to intel-mac-v3 in snd_hda_intel (sigmatel). Currently, even if you pass model=intel-mac-v3 as a module parameter to snd_hda_intel, the function patch_stac922x (patch_sigmatel.c) will still try to auto-detect the model type. This is a problem on my MacBook Pro 1st generation, which needs intel-mac-v3, but sometimes incorrectly reports 0x00000100 as subsystem id, which causes the switch in patch_stac922x to select intel-mac-v4. To fix this, I added a new model called intel-mac-auto, so in case no module parameter is passed, and an Intel Mac board is detected, the model will be automatically detected, while no detection will be done if the model is forced to intel-mac-v3. This problem has been around for quite a while, and I used to fix it by moving the case statement for 0x00000100 in patch_stac922x so that intel-mac-v3 is chosen. Another way to fix the problem would be to check if a module parameter was set directly in patch_stac922x, using something like this: if (spec->board_config == STAC_INTEL_MAC_V3 && !codec->bus->modelname) { But I think it is less elegant (if you prefer that way, I can prepare a patch). Signed-off-by: Nicolas Boichat Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_sigmatel.c | 14 +++++++++++--- 2 files changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 72aff61e7315..6f6d117ac7e2 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1024,6 +1024,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. intel-mac-v3 Intel Mac Type 3 intel-mac-v4 Intel Mac Type 4 intel-mac-v5 Intel Mac Type 5 + intel-mac-auto Intel Mac (detect type according to subsystem id) macmini Intel Mac Mini (equivalent with type 3) macbook Intel Mac Book (eq. type 5) macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 08cb77f51880..7fdafcb0015d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -94,6 +94,9 @@ enum { STAC_INTEL_MAC_V3, STAC_INTEL_MAC_V4, STAC_INTEL_MAC_V5, + STAC_INTEL_MAC_AUTO, /* This model is selected if no module parameter + * is given, one of the above models will be + * chosen according to the subsystem id. */ /* for backward compatibility */ STAC_MACMINI, STAC_MACBOOK, @@ -1483,6 +1486,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [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, + [STAC_INTEL_MAC_AUTO] = intel_mac_v3_pin_configs, /* for backward compatibility */ [STAC_MACMINI] = intel_mac_v3_pin_configs, [STAC_MACBOOK] = intel_mac_v5_pin_configs, @@ -1505,6 +1509,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_INTEL_MAC_V3] = "intel-mac-v3", [STAC_INTEL_MAC_V4] = "intel-mac-v4", [STAC_INTEL_MAC_V5] = "intel-mac-v5", + [STAC_INTEL_MAC_AUTO] = "intel-mac-auto", /* for backward compatibility */ [STAC_MACMINI] = "macmini", [STAC_MACBOOK] = "macbook", @@ -1576,9 +1581,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), /* other systems */ - /* Apple Mac Mini (early 2006) */ + /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, - "Mac Mini", STAC_INTEL_MAC_V3), + "Mac", STAC_INTEL_MAC_AUTO), /* Dell systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a7, "unknown Dell", STAC_922X_DELL_D81), @@ -3725,7 +3730,7 @@ 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_INTEL_MAC_V3) { + if (spec->board_config == STAC_INTEL_MAC_AUTO) { spec->gpio_mask = spec->gpio_dir = 0x03; spec->gpio_data = 0x03; /* Intel Macs have all same PCI SSID, so we need to check @@ -3757,6 +3762,9 @@ static int patch_stac922x(struct hda_codec *codec) case 0x106b2200: spec->board_config = STAC_INTEL_MAC_V5; break; + default: + spec->board_config = STAC_INTEL_MAC_V3; + break; } } -- cgit v1.2.3-58-ga151