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author | Linus Torvalds <torvalds@linux-foundation.org> | 2008-07-27 09:45:59 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2008-07-27 09:45:59 -0700 |
commit | 375614422509c98a1f3dbef410206bf81775169b (patch) | |
tree | 02e65184a80446d56b6c05b76417791a3b68b234 /sound | |
parent | eeb61f719c00c626115852bbc91189dc3011a844 (diff) | |
parent | 536319afd1f25383009c0c88f6fb00104f49c178 (diff) | |
download | blackbird-op-linux-375614422509c98a1f3dbef410206bf81775169b.tar.gz blackbird-op-linux-375614422509c98a1f3dbef410206bf81775169b.zip |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: Allow to force model to intel-mac-v3 in snd_hda_intel (sigmatel).
ALSA: cs4232: fix crash during chip PNP detection
ALSA: hda - Add automatic model setting for the Acer Aspire 5920G laptop
ALSA: make snd_ac97_add_vmaster() static
ALSA: sound/pci/azt3328.h: no variables for enums
ALSA: soc - wm9712 mono mixer
ALSA: hda - Add support of ASUS Eeepc P90*
ALSA: opti9xx: no isapnp param for !CONFIG_PNP
ALSA: opti93x - Fix NULL dereference
ALSA: hda - Added support for Asus V1Sn
ALSA: ASoC: Factor PGA DAPM handling into main
ALSA: ASoC: Refactor DAPM event handler
ALSA: ALSA: ens1370: communicate PCI device to AC97
ALSA: ens1370: SRC stands for Sample Rate Converter
ALSA: hda - Align BDL position adjustment parameter
ALSA: Au1xpsc: psc not disabled when TX is idle
ALSA: add TriTech 28023 AC97 codec ID and Wolfson 9701 name.
Diffstat (limited to 'sound')
-rw-r--r-- | sound/isa/cs423x/cs4236.c | 1 | ||||
-rw-r--r-- | sound/isa/opti9xx/opti92x-ad1848.c | 6 | ||||
-rw-r--r-- | sound/pci/ac97/ac97_codec.c | 3 | ||||
-rw-r--r-- | sound/pci/ac97/ac97_patch.c | 4 | ||||
-rw-r--r-- | sound/pci/azt3328.h | 4 | ||||
-rw-r--r-- | sound/pci/ens1370.c | 3 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 6 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 181 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 14 | ||||
-rw-r--r-- | sound/soc/au1x/psc-i2s.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm9712.c | 10 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 105 |
12 files changed, 259 insertions, 80 deletions
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; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 41c047e665ec..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); @@ -688,7 +692,7 @@ 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); } 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}, 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; 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 diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index fbf1124f7c79..9bf95367c882 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; } @@ -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; 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", diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2807bc840d26..add4e87e0b20 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 */ }; @@ -7905,6 +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), @@ -10946,7 +10949,23 @@ 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 +10987,27 @@ 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 +11023,13 @@ 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 +11037,7 @@ 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 +11100,98 @@ 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 +11323,9 @@ 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 +11353,16 @@ 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 +11371,32 @@ 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 +11450,6 @@ 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) @@ -12994,6 +13160,7 @@ 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), 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; } } 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)); 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"}, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061c2a6b..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) @@ -586,45 +568,56 @@ 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; + } + + /* 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)) { + 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; } } } |