From 1e1689536f346a431b748dc8ad9ac0828d2c065d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 08:34:32 +0200 Subject: ALSA: hda - Add missing static to patch_ca0110() Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 392d108c3558..019ca7cb56d7 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec) } -int patch_ca0110(struct hda_codec *codec) +static int patch_ca0110(struct hda_codec *codec) { struct ca0110_spec *spec; int err; -- cgit v1.2.2 From ff84847171508a3c76eb7e483204d1be7738729b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 18:08:01 +0200 Subject: ALSA: hda - Add quirk for HP 6930p Added a quirk model=laptop for HP 6930p (103c:30dc) with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 84cc49ca9148..85e8618e8497 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3966,6 +3966,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), -- cgit v1.2.2 From 826390796d09444b93e1f957582f8970ddfd9b3d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Jul 2009 08:31:30 +0200 Subject: sound: virtuoso: fix Xonar D1/DX silence after resume When resuming, we better take the DACs out of the reset state before trying to use them. Reference: kernel bug #13599 http://bugzilla.kernel.org/show_bug.cgi?id=13599 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bf971f7cfdc6..6ebcb6bdd712 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip) static void xonar_d1_resume(struct oxygen *chip) { + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); + msleep(1); cs43xx_init(chip); xonar_enable_output(chip); } -- cgit v1.2.2 From 099db17e66294b02814dee01c81d9abbbeece93e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Jul 2009 16:10:23 +0200 Subject: ALSA: hda - Add GPIO1 control at muting with HP laptops HP laptops with AD1984A codecs (at least mobile models) need to set GPIO1 appropriately to indicate the mute state. The BIOS checks this bit to judge whether the mute on or off is sent via F8 key. Without changing this bit, the BIOS can be confused and may toggle the mute wrongly. Reference: Novell bnc#515266 https://bugzilla.novell.com/show_bug.cgi?id=515266 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 85e8618e8497..f795ee588cc7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3734,9 +3734,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; +static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + int mute = (!ucontrol->value.integer.value[0] && + !ucontrol->value.integer.value[1]); + /* toggle GPIO1 according to the mute state */ + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + mute ? 0x02 : 0x0); + return ret; +} + static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), @@ -3857,6 +3878,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = { /* unsolicited event for pin-sense */ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ { } /* end */ }; -- cgit v1.2.2 From aa202455eec51699e44f658530728162cefa1307 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 15:00:54 +0200 Subject: ALSA: hda - Improve ASUS eeePC 1000 mixer The mixer elements created for ASUS eeePC 1000 with ALC269 aren't standard but strange words like "LineOut". Rename the element names to follow the standard one like "Headphone" and "Speaker". Also, split the volumes to each so that the virtual master can control them. The alc269_fujitsu_mixer is removed because it's now identical with the new eeepc mixer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a8e58c483df..e661b21354be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -/* 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), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { }; /* FSC amilo */ -static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol), - { } /* end */ -}; +#define alc269_fujitsu_mixer alc269_eeepc_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, -- cgit v1.2.2 From 022b466fc353d3dc7a152451144be656248666ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:03:30 +0200 Subject: ALSA: hda - Avoid invalid formats and rates with shared SPDIF Check whether formats and rates don't result in zero due to the restriction of SPDIF sharing. If any of them can be zero, disable the SPDIF sharing mode instead. Otherwise it will lead to a PCM configuration error. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2cedaa6a..26d255de6beb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } mutex_lock(&codec->spdif_mutex); if (mout->share_spdif) { - runtime->hw.rates &= mout->spdif_rates; - runtime->hw.formats &= mout->spdif_formats; - if (mout->spdif_maxbps < hinfo->maxbps) - hinfo->maxbps = mout->spdif_maxbps; + if ((runtime->hw.rates & mout->spdif_rates) && + (runtime->hw.formats & mout->spdif_formats)) { + runtime->hw.rates &= mout->spdif_rates; + runtime->hw.formats &= mout->spdif_formats; + if (mout->spdif_maxbps < hinfo->maxbps) + hinfo->maxbps = mout->spdif_maxbps; + } else { + mout->share_spdif = 0; + /* FIXME: need notify? */ + } } mutex_unlock(&codec->spdif_mutex); } -- cgit v1.2.2 From 70d321e6380f128096429d6e5b678f94ab0cef5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:06:45 +0200 Subject: ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback The PCM rates bit field may have been changed by the codec open callback. In that case, we need to reset rate_min and rate_max. So, simply call snd_pcm_lib_hw_rates() again after the codec open callback. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4e9ea7080270..b36dc46615a4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&chip->open_mutex); return err; } + snd_pcm_limit_hw_rates(runtime); spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; azx_dev->running = 0; -- cgit v1.2.2 From c470331e69bd54d11a9ea3c27a0e4ad783d02d6b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:10:23 +0200 Subject: ALSA: hda - Add sanity check in PCM open callback Add some sanity checks of struct snd_pcm_hardware fields in the PCM open callback of hda driver. This makes a bit easier to debug any PCM setup errors in the codec side. Signed-off-by: Takashi Iwai --- 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 b36dc46615a4..1877d95d4aa6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1464,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max)) + return -EINVAL; + if (snd_BUG_ON(!runtime->hw.formats)) + return -EINVAL; + if (snd_BUG_ON(!runtime->hw.rates)) + return -EINVAL; return 0; } -- cgit v1.2.2