From 3e3aecb4096a92f2dd8ac1abdabf4dd68218d67e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 13:37:50 +0100 Subject: ALSA: HDA: Enable surround and subwoofer on Lenovo Ideapad Y530 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit 32eea3884debb65ec1da633bc5df5aee23879865 upstream. The pin config values would change the association instead of the sequence, this commit fixes that up. Tested-by: Bartłomiej Żogała Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4261bb8eec1..8caf593fcf2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10748,6 +10748,7 @@ static struct alc_config_preset alc882_presets[] = { */ enum { PINFIX_ABIT_AW9D_MAX, + PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, }; @@ -10762,6 +10763,14 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [PINFIX_LENOVO_Y530] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x15, 0x99130112 }, /* rear int speakers */ + { 0x16, 0x99130111 }, /* subwoofer */ + { } + } + }, [PINFIX_PB_M5210] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10777,6 +10786,7 @@ static const struct alc_fixup alc882_fixups[] = { static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), {} -- cgit v1.2.3 From 8a805f1e7f3aea14039f7d57836e5f6ea993f37e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 14:08:30 +0100 Subject: ALSA: HDA: Fix volume control naming for surround speakers on Realtek auto-parser MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit ebbeb3d6aa22433c218da6f29fd7b3ebc89b87ea upstream. When more than one pair of internal speakers is present, allow names according to their channels. Tested-by: Bartłomiej Żogała Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8caf593fcf2..8c61566bdc5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5151,7 +5151,9 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, switch (cfg->line_out_type) { case AUTO_PIN_SPEAKER_OUT: - return "Speaker"; + if (cfg->line_outs == 1) + return "Speaker"; + break; case AUTO_PIN_HP_OUT: return "Headphone"; default: -- cgit v1.2.3 From f7ff876d0fba7e28c10c78a5004c350f816b8a6e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 14:22:25 +0100 Subject: ALSA: HDA: Fixup unnecessary volume control index on Realtek ALC88x MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4 upstream. Without this change, a volume control named "Surround" or "Side" would get an unnecessary index, causing it to be ignored by the vmaster and PulseAudio. Tested-by: Bartłomiej Żogała Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c61566bdc5..ffb05bd54ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5207,16 +5207,19 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) -- cgit v1.2.3 From 11052185a6eb08d7eb8b52a840f33c665226c8f1 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 16:54:52 +0100 Subject: ALSA: HDA: Realtek ALC88x: Do not over-initialize speakers and hp that are primary outputs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit 0a3fabe30e1a3b2037a12b863b8c45fffce38ee9 upstream. Do not initialize again the what has already been initialized as multi outs, as this breaks surround speakers. Tested-by: Bartłomiej Żogała Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 37 +++++++++++++++++++++---------------- 1 file changed, 21 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ffb05bd54ba..2a1c4124f30 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10844,23 +10844,28 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin, dac; int i; - for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { - pin = spec->autocfg.hp_pins[i]; - if (!pin) - break; - dac = spec->multiout.hp_nid; - if (!dac) - dac = spec->multiout.dac_nids[0]; /* to front */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + if (spec->autocfg.line_out_type != AUTO_PIN_HP_OUT) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + pin = spec->autocfg.hp_pins[i]; + if (!pin) + break; + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; /* to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } } - for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { - pin = spec->autocfg.speaker_pins[i]; - if (!pin) - break; - dac = spec->multiout.extra_out_nid[0]; - if (!dac) - dac = spec->multiout.dac_nids[0]; /* to front */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + + if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + pin = spec->autocfg.speaker_pins[i]; + if (!pin) + break; + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; /* to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } } -- cgit v1.2.3 From 6f1d6832db2fbade524f37cb23d246b45cb01e2a Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Wed, 9 Mar 2011 19:47:43 -0600 Subject: ALSA: hda - fix digital mic selection in mixer on 92HD8X codecs commit 094a42452abd5564429045e210281c6d22e67fca upstream. When the mux for digital mic is different from the mux for other mics, the current auto-parser doesn't handle them in a right way but provides only one mic. This patch fixes the issue. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_sigmatel.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bd7b123f644..052062dd523 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -757,7 +757,7 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct sigmatel_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); const struct hda_input_mux *imux = spec->input_mux; - unsigned int idx, prev_idx; + unsigned int idx, prev_idx, didx; idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) @@ -769,7 +769,8 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_CONNECT_SEL, imux->items[idx].index); - if (prev_idx >= spec->num_analog_muxes) { + if (prev_idx >= spec->num_analog_muxes && + spec->mux_nids[adc_idx] != spec->dmux_nids[adc_idx]) { imux = spec->dinput_mux; /* 0 = analog */ snd_hda_codec_write_cache(codec, @@ -779,9 +780,13 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e } } else { imux = spec->dinput_mux; + /* first dimux item is hardcoded to select analog imux, + * so lets skip it + */ + didx = idx - spec->num_analog_muxes + 1; snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0, AC_VERB_SET_CONNECT_SEL, - imux->items[idx - 1].index); + imux->items[didx].index); } spec->cur_mux[adc_idx] = idx; return 1; -- cgit v1.2.3 From 24449f05b663f1984ee6ecdba5a0d2ff2a1e0462 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Mar 2011 12:51:11 +0100 Subject: ALSA: hda - Initialize special cases for input src in init phase commit 584c0c4c359bdac37d94157f8d7fc513d26c8328 upstream. Currently some special handling for the unusual case like dual-ADCs or a single-input-src is done in the tree-parse time in set_capture_mixer(). But this setup could be overwritten by static init verbs. This patch moves the initialization into the init phase so that such input-src setup won't be lost. Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2a1c4124f30..acd209958ea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -394,6 +394,7 @@ struct alc_spec { /* other flags */ unsigned int no_analog :1; /* digital I/O only */ unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */ + unsigned int single_input_src:1; int init_amp; int codec_variant; /* flag for other variants */ @@ -3919,6 +3920,8 @@ static struct hda_amp_list alc880_lg_loopbacks[] = { * Common callbacks */ +static void alc_init_special_input_src(struct hda_codec *codec); + static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -3929,6 +3932,7 @@ static int alc_init(struct hda_codec *codec) for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + alc_init_special_input_src(codec); if (spec->init_hook) spec->init_hook(codec); @@ -5590,6 +5594,7 @@ static void fixup_single_adc(struct hda_codec *codec) spec->capsrc_nids += i; spec->adc_nids += i; spec->num_adc_nids = 1; + spec->single_input_src = 1; } } @@ -5601,6 +5606,16 @@ static void fixup_dual_adc_switch(struct hda_codec *codec) init_capsrc_for_pin(codec, spec->int_mic.pin); } +/* initialize some special cases for input sources */ +static void alc_init_special_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + if (spec->dual_adc_switch) + fixup_dual_adc_switch(codec); + else if (spec->single_input_src) + init_capsrc_for_pin(codec, spec->autocfg.inputs[0].pin); +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5616,7 +5631,7 @@ static void set_capture_mixer(struct hda_codec *codec) int mux = 0; int num_adcs = spec->num_adc_nids; if (spec->dual_adc_switch) - fixup_dual_adc_switch(codec); + num_adcs = 1; else if (spec->auto_mic) fixup_automic_adc(codec); else if (spec->input_mux) { @@ -5625,8 +5640,6 @@ static void set_capture_mixer(struct hda_codec *codec) else if (spec->input_mux->num_items == 1) fixup_single_adc(codec); } - if (spec->dual_adc_switch) - num_adcs = 1; spec->cap_mixer = caps[mux][num_adcs - 1]; } } -- cgit v1.2.3 From f97aa1f43ed9e241a05f7e4d2e4afedf97356f7a Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Thu, 17 Mar 2011 18:32:24 -0400 Subject: ALSA: sound/pci/asihpi: check adapter index in hpi_ioctl commit 4a122c10fbfe9020df469f0f669da129c5757671 upstream. The user-supplied index into the adapters array needs to be checked, or an out-of-bounds kernel pointer could be accessed and used, leading to potentially exploitable memory corruption. Signed-off-by: Dan Rosenberg Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/asihpi/hpioctl.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 22dbd91811a..448dd01943f 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -155,6 +155,11 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } + if (hm->h.adapter_index >= HPI_MAX_ADAPTERS) { + err = -EINVAL; + goto out; + } + pa = &adapters[hm->h.adapter_index]; hr->h.size = 0; if (hm->h.object == HPI_OBJ_SUBSYSTEM) { -- cgit v1.2.3 From 74d5036bb343ce72af45f8bdcbd5dd87459d1216 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Mar 2011 07:31:53 +0100 Subject: ALSA: aloop - Fix possible IRQ lock inversion commit 98d21df431ad55281e1abf780f8d51e3391900b2 upstream. loopback_pos_update() can be called in the timer callback, thus the lock held should be irq-safe. Otherwise you'll get AB/BA deadlock together with substream->self_group.lock. Reported-and-tested-by: Knut Petersen Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/drivers/aloop.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 12b44b0b677..a0da7755fce 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -482,8 +482,9 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) cable->streams[SNDRV_PCM_STREAM_CAPTURE]; unsigned long delta_play = 0, delta_capt = 0; unsigned int running; + unsigned long flags; - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { delta_play = jiffies - dpcm_play->last_jiffies; @@ -495,10 +496,8 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) dpcm_capt->last_jiffies += delta_capt; } - if (delta_play == 0 && delta_capt == 0) { - spin_unlock(&cable->lock); - return running; - } + if (delta_play == 0 && delta_capt == 0) + goto unlock; if (delta_play > delta_capt) { loopback_bytepos_update(dpcm_play, delta_play - delta_capt, @@ -510,14 +509,14 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) delta_capt = delta_play; } - if (delta_play == 0 && delta_capt == 0) { - spin_unlock(&cable->lock); - return running; - } + if (delta_play == 0 && delta_capt == 0) + goto unlock; + /* note delta_capt == delta_play at this moment */ loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY); loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY); - spin_unlock(&cable->lock); + unlock: + spin_unlock_irqrestore(&cable->lock, flags); return running; } -- cgit v1.2.3 From 5794354c837af7dbbe7793d689198c446ab68bfd Mon Sep 17 00:00:00 2001 From: Przemyslaw Bruski Date: Sun, 13 Mar 2011 16:18:56 +0100 Subject: ALSA: ctxfi - Fix incorrect SPDIF status bit mask commit 4c1847e884efddcc3ede371f7839e5e65b25c34d upstream. SPDIF status mask creation was incorrect. Signed-off-by: Przemyslaw Bruski Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/ctxfi/ctatc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 1bff80cde0a..b9321544c31 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -869,7 +869,7 @@ spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm *apcm) mutex_lock(&atc->atc_mutex); dao->ops->get_spos(dao, &status); if (((status >> 24) & IEC958_AES3_CON_FS) != iec958_con_fs) { - status &= ((~IEC958_AES3_CON_FS) << 24); + status &= ~(IEC958_AES3_CON_FS << 24); status |= (iec958_con_fs << 24); dao->ops->set_spos(dao, status); dao->ops->commit_write(dao); -- cgit v1.2.3 From f13d689d83102b120af29b659ade12da22ab717f Mon Sep 17 00:00:00 2001 From: Przemyslaw Bruski Date: Sun, 13 Mar 2011 16:18:57 +0100 Subject: ALSA: ctxfi - Fix SPDIF status retrieval commit f164753a263bfd2daaf3e0273b179de7e099c57d upstream. SDPIF status retrieval always returned the default settings instead of the actual ones. Signed-off-by: Przemyslaw Bruski Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/ctxfi/ctmixer.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 15c1e7271ea..c3519ff42fb 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -566,19 +566,6 @@ static int ct_spdif_get_mask(struct snd_kcontrol *kcontrol, return 0; } -static int ct_spdif_default_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int status = SNDRV_PCM_DEFAULT_CON_SPDIF; - - ucontrol->value.iec958.status[0] = (status >> 0) & 0xff; - ucontrol->value.iec958.status[1] = (status >> 8) & 0xff; - ucontrol->value.iec958.status[2] = (status >> 16) & 0xff; - ucontrol->value.iec958.status[3] = (status >> 24) & 0xff; - - return 0; -} - static int ct_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -586,6 +573,10 @@ static int ct_spdif_get(struct snd_kcontrol *kcontrol, unsigned int status; atc->spdif_out_get_status(atc, &status); + + if (status == 0) + status = SNDRV_PCM_DEFAULT_CON_SPDIF; + ucontrol->value.iec958.status[0] = (status >> 0) & 0xff; ucontrol->value.iec958.status[1] = (status >> 8) & 0xff; ucontrol->value.iec958.status[2] = (status >> 16) & 0xff; @@ -629,7 +620,7 @@ static struct snd_kcontrol_new iec958_default_ctl = { .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .count = 1, .info = ct_spdif_info, - .get = ct_spdif_default_get, + .get = ct_spdif_get, .put = ct_spdif_put, .private_value = MIXER_IEC958_DEFAULT }; -- cgit v1.2.3 From 44154ce1db1501e54a0da6f6ad2d49c9c2ed3bb0 Mon Sep 17 00:00:00 2001 From: Przemyslaw Bruski Date: Sun, 13 Mar 2011 16:18:58 +0100 Subject: ALSA: ctxfi - Clear input settings before initialization commit efed5f26664f93991c929d5bb343e65f900d72bc upstream. Clear input settings before initialization. Signed-off-by: Przemyslaw Bruski Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/ctxfi/ctdaio.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index af56eb949bd..47d9ea97de0 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -176,6 +176,7 @@ static int dao_set_left_input(struct dao *dao, struct rsc *input) if (!entry) return -ENOMEM; + dao->ops->clear_left_input(dao); /* Program master and conjugate resources */ input->ops->master(input); daio->rscl.ops->master(&daio->rscl); @@ -204,6 +205,7 @@ static int dao_set_right_input(struct dao *dao, struct rsc *input) if (!entry) return -ENOMEM; + dao->ops->clear_right_input(dao); /* Program master and conjugate resources */ input->ops->master(input); daio->rscr.ops->master(&daio->rscr); -- cgit v1.2.3 From 1502ccf37a1cafc8ff1f45f8999854b6210abf91 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 11 Mar 2011 11:05:38 +0200 Subject: ASoC: PXA: Z2: Fix codec pin name commit 64c25a92e865f06ad8782fbdaa1e2a97d50acf73 upstream. MONO was renamed to MONO1. Signed-off-by: Vasily Khoruzhick Acked-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/pxa/z2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 3ceaef68e01..838a0d540c5 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -147,7 +147,7 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "LINPUT3"); snd_soc_dapm_disable_pin(dapm, "RINPUT3"); snd_soc_dapm_disable_pin(dapm, "OUT3"); - snd_soc_dapm_disable_pin(dapm, "MONO"); + snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Add z2 specific widgets */ snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, -- cgit v1.2.3