From b3df026ea230b233f5a4ebc7400033f7326fad12 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Feb 2013 23:35:46 +0000 Subject: ASoC: wm8960: Correct register 0 and 1 defaults Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb9273..4cb49eb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -53,8 +53,8 @@ * using 2 wire for device control, so we cache them instead. */ static const struct reg_default wm8960_reg_defaults[] = { - { 0x0, 0x0097 }, - { 0x1, 0x0097 }, + { 0x0, 0x00a7 }, + { 0x1, 0x00a7 }, { 0x2, 0x0000 }, { 0x3, 0x0000 }, { 0x4, 0x0000 }, -- cgit v1.1 From 44426de4d87682870b35a649b76586177113f5e7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Feb 2013 23:36:48 +0000 Subject: ASoC: wm8960: Fix ADC power bits Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4cb49eb..a64b934 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -323,8 +323,8 @@ SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0, SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0, wm8960_rin, ARRAY_SIZE(wm8960_rin)), -SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0), -SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0), +SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER1, 3, 0), +SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0), SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), -- cgit v1.1 From c9b5669031a72026e41c4a7a094ac1efa4ef0ef5 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 15 Feb 2013 10:37:26 +0000 Subject: ASoC: wm5102: Correct OUT2 volume and switch names Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ab69c83..deb1097 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -755,7 +755,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), @@ -767,7 +767,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, -- cgit v1.1 From 5752ec93f3a120f0d4088565989eaea27db7a0d8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 15 Feb 2013 10:37:27 +0000 Subject: ASoC: wm5110: Correct OUT2/3 volume and switch names Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index a163132..1a97263 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -213,9 +213,9 @@ ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, ARIZONA_OUT1_OSR_SHIFT, 1, 0), -SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, +SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, ARIZONA_OUT2_OSR_SHIFT, 1, 0), -SOC_SINGLE("OUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, +SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, ARIZONA_OUT3_OSR_SHIFT, 1, 0), SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, ARIZONA_OUT4_OSR_SHIFT, 1, 0), @@ -226,9 +226,9 @@ SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, +SOC_DOUBLE_R("HPOUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), @@ -240,10 +240,10 @@ SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, +SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, @@ -260,11 +260,11 @@ SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, ARIZONA_OUTPUT_PATH_CONFIG_1R, ARIZONA_OUT1L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, +SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, ARIZONA_OUTPUT_PATH_CONFIG_2R, ARIZONA_OUT2L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("OUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, +SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, ARIZONA_OUTPUT_PATH_CONFIG_3R, ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), -- cgit v1.1 From 51cd02d43c0bc7f55c84f50de23c08554db56ce1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 3 Mar 2013 16:20:50 +0800 Subject: ASoC: wm8350: Use jiffies rather than msecs in schedule_delayed_work() The delay parameter of schedule_delayed_work() is number of jiffies to wait rather than miliseconds. Before commit 6d3c26bcb "ASoC: Use delayed work to debounce WM8350 jack IRQs", the debounce time is 200 miliseconds in wm8350_hp_jack_handler(). So I think this is a bug when convert to use delayed work. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index fb92fb4..1db957d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1303,7 +1303,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, 200); + schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1320,7 +1320,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, 200); + schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.1 From 85c50a5899b23f4f893b0898b286023157b98376 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 4 Mar 2013 15:19:18 +0300 Subject: ALSA: seq: seq_oss_event: missing range checks The "dev" variable could be out of bounds. Calling snd_seq_oss_synth_is_valid() checks that it is is a valid device which has been opened. We check this inside set_note_event() so this function can't succeed without a valid "dev". But we need to do the check earlier to prevent invalid dereferences and memory corruption. One call tree where "dev" could be out of bounds is: -> snd_seq_oss_oob_user() -> snd_seq_oss_process_event() -> extended_event() -> note_on_event() Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_event.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 066f5f3..c390886 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev static int note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { @@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st static int note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { -- cgit v1.1 From 0af18c5cc9403999bb189f825b816f7fc80fc0ee Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 4 Mar 2013 17:10:20 -0700 Subject: ASoC: tegra: fix I2S bit count mask This register field is 11 bits wide, not 15 bits wide. Given the way this value is currently, used, this patch has no practical effect. However, it's still best if the value is correct. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_i2s.h | 2 +- sound/soc/tegra/tegra30_i2s.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index c27069d..7299587 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -121,7 +121,7 @@ #define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA20_I2S_FIFO_SCR */ diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 34dc47b..a294d94 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -110,7 +110,7 @@ #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA30_I2S_OFFSET */ -- cgit v1.1 From 2069d483b39a603a5f3428a19d3b4ac89aa97f48 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Mar 2013 15:43:39 +0100 Subject: ALSA: vmaster: Fix slave change notification When a value of a vmaster slave control is changed, the ctl change notification is sometimes ignored. This happens when the master control overrides, e.g. when the corresponding master control is muted. The reason is that slave_put() returns the value of the actual slave put callback, and it doesn't reflect the virtual slave value change. This patch fixes the function just to return 1 whenever a slave value is changed. Cc: Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 8575861..0097f36 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol, } if (!changed) return 0; - return slave_put_val(slave, ucontrol); + err = slave_put_val(slave, ucontrol); + if (err < 0) + return err; + return 1; } static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 82968b7e8d6150fcea0b48488f7bf6fb25e7b099 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Mar 2013 22:59:35 +0800 Subject: ASoC: wm5102: Apply a SYSCLK patch for later revs Evaluation has identified some performance improvements to the device. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index deb1097..cef288c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -573,6 +573,13 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = { { 0x025e, 0x0112 }, }; +static const struct reg_default wm5102_sysclk_revb_patch[] = { + { 0x3081, 0x08FE }, + { 0x3083, 0x00ED }, + { 0x30C1, 0x08FE }, + { 0x30C3, 0x00ED }, +}; + static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, patch = wm5102_sysclk_reva_patch; patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); break; + default: + patch = wm5102_sysclk_revb_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch); + break; } switch (event) { -- cgit v1.1 From 25336e8ae2d2fa64c9c4cc2c9c28f641134c9fa9 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 7 Mar 2013 14:10:25 -0500 Subject: ALSA: hda - check NULL pointer when creating SPDIF controls If the SPDIF control array cannot be reallocated, the function will return to avoid dereferencing a NULL pointer. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..3dc6566 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3334,6 +3334,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, return -EBUSY; } spdif = snd_array_new(&codec->spdif_out); + if (!spdif) + return -ENOMEM; for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); if (!kctl) -- cgit v1.1 From 4c7a548a70a44269266858f65c3b5fc9c3ace057 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 7 Mar 2013 14:11:05 -0500 Subject: ALSA: hda - check NULL pointer when creating SPDIF PCM switch If the new control cannot be created, this function will return to avoid snd_hda_ctl_add dereferencing a NULL control pointer. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3dc6566..97c68dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3433,11 +3433,16 @@ static struct snd_kcontrol_new spdif_share_sw = { int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { + struct snd_kcontrol *kctl; + if (!mout->dig_out_nid) return 0; + + kctl = snd_ctl_new1(&spdif_share_sw, mout); + if (!kctl) + return -ENOMEM; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, mout->dig_out_nid, - snd_ctl_new1(&spdif_share_sw, mout)); + return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl); } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); -- cgit v1.1 From 3bc085a12d8f9f3e45a4ac0cc24a34abd5b20657 Mon Sep 17 00:00:00 2001 From: Xi Wang Date: Thu, 7 Mar 2013 00:13:51 -0500 Subject: ALSA: hda/ca0132 - Avoid division by zero in dspxfr_one_seg() Move the zero check `hda_frame_size_words == 0' before the modulus `buffer_size_words % hda_frame_size_words'. Also remove the redundant null check `buffer_addx == NULL'. Signed-off-by: Xi Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index db02c1e..eefc456 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec, hda_frame_size_words = ((sample_rate_div == 0) ? 0 : (num_chans * sample_rate_mul / sample_rate_div)); + if (hda_frame_size_words == 0) { + snd_printdd(KERN_ERR "frmsz zero\n"); + return -EINVAL; + } + buffer_size_words = min(buffer_size_words, (unsigned int)(UC_RANGE(chip_addx, 1) ? 65536 : 32768)); @@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, chip_addx, hda_frame_size_words, num_chans, sample_rate_mul, sample_rate_div, buffer_size_words); - if ((buffer_addx == NULL) || (hda_frame_size_words == 0) || - (buffer_size_words < hda_frame_size_words)) { + if (buffer_size_words < hda_frame_size_words) { snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n"); return -EINVAL; } -- cgit v1.1 From 84dfd0ac231f69d70e100e712ad5e5f0092ad46b Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 7 Mar 2013 09:19:38 +0100 Subject: ALSA: hda - Add support of new codec ALC233 It's compatible with ALC282. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d4237b..563c24d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3163,6 +3163,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0290: spec->codec_variant = ALC269_TYPE_ALC280; break; + case 0x10ec0233: case 0x10ec0282: case 0x10ec0283: spec->codec_variant = ALC269_TYPE_ALC282; @@ -3862,6 +3863,7 @@ static int patch_alc680(struct hda_codec *codec) */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, + { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, -- cgit v1.1 From 69a4cfdd444d1fe5c24d29b3a063964ac165d2cd Mon Sep 17 00:00:00 2001 From: Sean Connor Date: Thu, 28 Feb 2013 09:20:00 -0500 Subject: ALSA: ice1712: Initialize card->private_data properly Set card->private_data in snd_ice1712_create for fixing NULL dereference in snd_ice1712_remove(). Signed-off-by: Sean Connor Cc: Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 2ffdc35..806407a 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2594,6 +2594,8 @@ static int snd_ice1712_create(struct snd_card *card, snd_ice1712_proc_init(ice); synchronize_irq(pci->irq); + card->private_data = ice; + err = pci_request_regions(pci, "ICE1712"); if (err < 0) { kfree(ice); -- cgit v1.1 From 66efdc71d95887b652a742a5dae51fa834d71465 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Mar 2013 18:11:17 +0100 Subject: ALSA: seq: Fix missing error handling in snd_seq_timer_open() snd_seq_timer_open() didn't catch the whole error path but let through if the timer id is a slave. This may lead to Oops by accessing the uninitialized pointer. BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae IP: [] snd_seq_timer_open+0xe7/0x130 PGD 785cd067 PUD 76964067 PMD 0 Oops: 0002 [#4] SMP CPU 0 Pid: 4288, comm: trinity-child7 Tainted: G D W 3.9.0-rc1+ #100 Bochs Bochs RIP: 0010:[] [] snd_seq_timer_open+0xe7/0x130 RSP: 0018:ffff88006ece7d38 EFLAGS: 00010246 RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000 RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38 RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007 FS: 00007f7aaa650700(0000) GS:ffff88007f800000(0000) GS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290) Stack: 0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d 65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000 ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520 Call Trace: [] ? trace_hardirqs_on+0xd/0x10 [] snd_seq_queue_timer_open+0x29/0x70 [] snd_seq_ioctl_set_queue_timer+0xda/0x120 [] snd_seq_do_ioctl+0x9b/0xd0 [] snd_seq_ioctl+0x10/0x20 [] do_vfs_ioctl+0x522/0x570 [] ? file_has_perm+0x83/0xa0 [] ? trace_hardirqs_on+0xd/0x10 [] sys_ioctl+0x5d/0xa0 [] ? trace_hardirqs_on_thunk+0x3a/0x3f [] system_call_fastpath+0x16/0x1b Reported-and-tested-by: Tommi Rantala Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_timer.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 160b1bd..24d44b2 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -290,10 +290,10 @@ int snd_seq_timer_open(struct snd_seq_queue *q) tid.device = SNDRV_TIMER_GLOBAL_SYSTEM; err = snd_timer_open(&t, str, &tid, q->queue); } - if (err < 0) { - snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err); - return err; - } + } + if (err < 0) { + snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err); + return err; } t->callback = snd_seq_timer_interrupt; t->callback_data = q; -- cgit v1.1 From 2e9b9a3c243b1bc1fc9d1e84fcbc32568467bf8e Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 12 Mar 2013 00:16:40 +0800 Subject: ALSA: asihpi - fix potential NULL pointer dereference The dereference should be moved below the NULL test. Signed-off-by: Wei Yongjun Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 3536b07..0aabfed 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi, static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) { - struct snd_card *card = asihpi->card; + struct snd_card *card; unsigned int idx = 0; unsigned int subindex = 0; int err; @@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) if (snd_BUG_ON(!asihpi)) return -EINVAL; + card = asihpi->card; strcpy(card->mixername, "Asihpi Mixer"); err = -- cgit v1.1 From 281a6ac0f54052c81bbee156914459ba5a08f924 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 11 Mar 2013 20:15:34 +0100 Subject: ALSA: usb-audio: add a workaround for the NuForce UDH-100 The NuForce UDH-100 numbers its interfaces incorrectly, which makes the interface associations come out wrong, which results in the driver erroring out with the message "Audio class v2 interfaces need an interface association". Work around this by searching for the interface association descriptor also in some other place where it might have ended up. Reported-and-tested-by: Dave Helstroom Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 803953a..2da8ad7 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -244,6 +244,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) usb_ifnum_to_if(dev, ctrlif)->intf_assoc; if (!assoc) { + /* + * Firmware writers cannot count to three. So to find + * the IAD on the NuForce UDH-100, also check the next + * interface. + */ + struct usb_interface *iface = + usb_ifnum_to_if(dev, ctrlif + 1); + if (iface && + iface->intf_assoc && + iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO && + iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2) + assoc = iface->intf_assoc; + } + + if (!assoc) { snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); return -EINVAL; } -- cgit v1.1 From b5f82b1044daef74059f454353a2ee97acbbe620 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Mar 2013 16:47:30 +0100 Subject: ALSA: hda - Fix snd_hda_get_num_raw_conns() to return a correct value In the connection list expansion in hda_codec.c and hda_proc.c, the value returned from snd_hda_get_num_raw_conns() is used as the array size to store the connection list. However, the function returns simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the widget list with ranges isn't considered there. Thus it may return a smaller size than the actual list, which results in -ENOSPC in snd_hda_get_raw_conections(). This patch fixes the bug by parsing the connection list correctly also for snd_hda_get_num_raw_conns(). Reported-and-tested-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 97c68dd..a9ebcf9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -494,7 +494,7 @@ static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - return get_num_conns(codec, nid) & AC_CLIST_LENGTH; + return snd_hda_get_raw_connections(codec, nid, NULL, 0); } /** @@ -517,9 +517,6 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t prev_nid; int null_count = 0; - if (snd_BUG_ON(!conn_list || max_conns <= 0)) - return -EINVAL; - parm = get_num_conns(codec, nid); if (!parm) return 0; @@ -545,7 +542,8 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, AC_VERB_GET_CONNECT_LIST, 0); if (parm == -1 && codec->bus->rirb_error) return -EIO; - conn_list[0] = parm & mask; + if (conn_list) + conn_list[0] = parm & mask; return 1; } @@ -580,14 +578,20 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { + if (conn_list) { + if (conns >= max_conns) + return -ENOSPC; + conn_list[conns] = n; + } + conns++; + } + } else { + if (conn_list) { if (conns >= max_conns) return -ENOSPC; - conn_list[conns++] = n; + conn_list[conns] = val; } - } else { - if (conns >= max_conns) - return -ENOSPC; - conn_list[conns++] = val; + conns++; } prev_nid = val; } -- cgit v1.1 From 303985f81019571db0b3a6f01fc7f03eb350657e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 14 Mar 2013 15:28:29 +0100 Subject: ALSA: hda - Disable IDT eapd_switch if there are no internal speakers If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 83d5335..dafe04a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) return 0; } +/* check whether a built-in speaker is included in parsed pins */ +static bool has_builtin_speaker(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t *nid_pin; + int nids, i; + + if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) { + nid_pin = spec->gen.autocfg.line_out_pins; + nids = spec->gen.autocfg.line_outs; + } else { + nid_pin = spec->gen.autocfg.speaker_pins; + nids = spec->gen.autocfg.speaker_outs; + } + + for (i = 0; i < nids; i++) { + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]); + if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT) + return true; + } + return false; +} + /* * PC beep controls */ @@ -3890,6 +3913,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; } + /* Don't GPIO-mute speakers if there are no internal speakers, because + * the GPIO might be necessary for Headphone + */ + if (spec->eapd_switch && !has_builtin_speaker(codec)) + spec->eapd_switch = 0; + codec->proc_widget_hook = stac92hd7x_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); -- cgit v1.1 From d1d28500cccc269fdbf81ba33d7328d1d2c04b2f Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 14 Mar 2013 17:27:44 -0700 Subject: ALSA: hda/ca0132 - Check if dspload_image succeeded. If dspload_image() fails, it was ignored and dspload_wait_loaded() was still called. dsp_loaded should never be set to true in this case, skip it. The check in dspload_wait_loaded() return true if the DSP is loaded or if it never started. Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index eefc456..cf24b75 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4351,12 +4351,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) return false; dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); - dspload_image(codec, dsp_os_image, 0, 0, true, 0); + if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { + pr_err("ca0132 dspload_image failed.\n"); + goto exit_download; + } + dsp_loaded = dspload_wait_loaded(codec); +exit_download: release_firmware(fw_entry); - return dsp_loaded; } -- cgit v1.1 From e8f1bd5d77484a1088797fd5689b1a37148a170e Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 14 Mar 2013 17:27:45 -0700 Subject: ALSA: hda/ca0132 - Check download state of DSP. Instead of using the dspload_is_loaded() function, check the dsp_state that is kept in the spec. The dspload_is_loaded() function returns true if the DSP transfer was never started. This false-positive leads to multiple second delays when ca0132_setup_efaults() times out on each write. Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index cf24b75..225d1d5 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3239,7 +3239,7 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val) struct ca0132_spec *spec = codec->spec; unsigned int tmp; - if (!dspload_is_loaded(codec)) + if (spec->dsp_state != DSP_DOWNLOADED) return 0; /* if CrystalVoice if off, vipsource should be 0 */ @@ -4267,11 +4267,12 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) */ static void ca0132_setup_defaults(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; unsigned int tmp; int num_fx; int idx, i; - if (!dspload_is_loaded(codec)) + if (spec->dsp_state != DSP_DOWNLOADED) return; /* out, in effects + voicefx */ -- cgit v1.1 From b714a7106ba5423c418c25e6231116560f8a9ef8 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 14 Mar 2013 17:27:46 -0700 Subject: ALSA: hda/ca0132 - Remove extra setting of dsp_state. spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset and check it in ca0132_download_dsp(). Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 225d1d5..0792b57 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4372,16 +4372,13 @@ static void ca0132_download_dsp(struct hda_codec *codec) #ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP return; /* NOP */ #endif - spec->dsp_state = DSP_DOWNLOAD_INIT; - if (spec->dsp_state == DSP_DOWNLOAD_INIT) { - chipio_enable_clocks(codec); - spec->dsp_state = DSP_DOWNLOADING; - if (!ca0132_download_dsp_images(codec)) - spec->dsp_state = DSP_DOWNLOAD_FAILED; - else - spec->dsp_state = DSP_DOWNLOADED; - } + chipio_enable_clocks(codec); + spec->dsp_state = DSP_DOWNLOADING; + if (!ca0132_download_dsp_images(codec)) + spec->dsp_state = DSP_DOWNLOAD_FAILED; + else + spec->dsp_state = DSP_DOWNLOADED; if (spec->dsp_state == DSP_DOWNLOADED) ca0132_set_dsp_msr(codec, true); -- cgit v1.1 From 57220bc1f5924c869d8fc049e50169915ca0cb24 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 15 Mar 2013 09:14:22 +0300 Subject: sound: sequencer: cap array index in seq_chn_common_event() "chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sequencer.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 30bcfe4..4ff60a6 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec) case MIDI_PGM_CHANGE: if (seq_mode == SEQ_2) { + if (chn > 15) + break; + synth_devs[dev]->chn_info[chn].pgm_num = p1; if ((int) dev >= num_synths) synth_devs[dev]->set_instr(dev, chn, p1); @@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec) case MIDI_PITCH_BEND: if (seq_mode == SEQ_2) { + if (chn > 15) + break; + synth_devs[dev]->chn_info[chn].bender_value = w14; if ((int) dev < num_synths) -- cgit v1.1 From 6d3073e124e1a6138b929479301d3a7ecde00f27 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Mar 2013 14:23:32 +0100 Subject: ALSA: hda - Fix missing EAPD/GPIO setup for Cirrus codecs During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 72ebb8a..60d08f6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -506,6 +506,8 @@ static int patch_cs420x(struct hda_codec *codec) if (!spec) return -ENOMEM; + spec->gen.automute_hook = cs_automute; + snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, cs420x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -893,6 +895,8 @@ static int patch_cs4210(struct hda_codec *codec) if (!spec) return -ENOMEM; + spec->gen.automute_hook = cs_automute; + snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl, cs421x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); -- cgit v1.1 From 31b6945a899a30f9dffa9cba8ed2e494784810a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Mar 2013 10:23:40 +0100 Subject: ALSA: hda - Fix missing beep detach in patch_conexant.c This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 941bf6c..1051a88 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec) return 0; } +static void cx_auto_free(struct hda_codec *codec) +{ + snd_hda_detach_beep_device(codec); + snd_hda_gen_free(codec); +} + static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, - .free = snd_hda_gen_free, + .free = cx_auto_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, -- cgit v1.1 From a86b1a2cd2f81f74e815e07f756edd7bc5b6f034 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Mar 2013 11:00:44 +0100 Subject: ALSA: hda/cirrus - Fix the digital beep registration The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1051a88..2a89d1ee 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -3397,7 +3397,7 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->patch_ops = cx_auto_patch_ops; if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. -- cgit v1.1 From 039eb75350acd1131a18a9bd12a0d4e1fb17892e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Mar 2013 16:55:49 +0100 Subject: ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver I forgot to update spec->gpio_data in the automute hook, so it will be overridden at the init sequence, thus the machine is still silent when no headphone jack is plugged at boot time. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 60d08f6..0d9c58f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -168,10 +168,10 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); if (spec->gpio_eapd_hp) { - unsigned int gpio = spec->gen.hp_jack_present ? + spec->gpio_data = spec->gen.hp_jack_present ? spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, gpio); + AC_VERB_SET_GPIO_DATA, spec->gpio_data); } } -- cgit v1.1 From 61ac51301e6c6d4ed977d7674ce2b8e713619a9b Mon Sep 17 00:00:00 2001 From: Torstein Hegge Date: Tue, 19 Mar 2013 17:12:14 +0100 Subject: ALSA: usb: Parse UAC2 extension unit like for UAC1 UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge Acked-by: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f7..8eb84c0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -725,7 +725,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC1_PROCESSING_UNIT: case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ { + /* UAC2_EFFECT_UNIT */ + case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; if (state->mixer->protocol == UAC_VERSION_2 && @@ -2052,6 +2053,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_extension_unit(state, unitid, p1); else /* UAC_VERSION_2 */ return parse_audio_processing_unit(state, unitid, p1); + case UAC2_EXTENSION_UNIT_V2: + return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); return -EINVAL; -- cgit v1.1 From 4d7b86c98e445b075c2c4c3757eb6d3d6efbe72e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 19 Mar 2013 21:09:24 +0100 Subject: ALSA: snd-usb: mixer: propagate errors up the call chain In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 8eb84c0..45cc0af 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, d->baSourceID[0], term) < 0) - return -ENODEV; + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); @@ -1357,8 +1358,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return err; /* determine the input source type and name */ - if (check_input_term(state, hdr->bSourceID, &iterm) < 0) - return -EINVAL; + err = check_input_term(state, hdr->bSourceID, &iterm); + if (err < 0) + return err; master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ -- cgit v1.1 From 83ea5d18d74f032a760fecde78c0210f66f7f70c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 19 Mar 2013 21:09:25 +0100 Subject: ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls() Creation of individual mixer controls may fail, but that shouldn't cause the entire mixer creation to fail. Even worse, if the mixer creation fails, that will error out the entire device probing. All the functions called by parse_audio_unit() should return -EINVAL if they find descriptors that are unsupported or believed to be malformed, so we can safely handle this error code as a non-fatal condition in snd_usb_mixer_controls(). That fixes a long standing bug which is commonly worked around by adding quirks which make the driver ignore entire interfaces. Some of them might now be unnecessary. Signed-off-by: Daniel Mack Reported-and-tested-by: Rodolfo Thomazelli Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 45cc0af..ca4739c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2123,7 +2123,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; } else { /* UAC_VERSION_2 */ struct uac2_output_terminal_descriptor *desc = p; @@ -2135,12 +2135,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; /* for UAC2, use the same approach to also add the clock selectors */ err = parse_audio_unit(&state, desc->bCSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; } } -- cgit v1.1 From a686fd141e20244ad75f80ad54706da07d7bb90a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Mar 2013 15:42:00 +0100 Subject: ALSA: hda - Fix typo in checking IEC958 emphasis bit There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a9ebcf9..ecdf30e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3144,7 +3144,7 @@ static unsigned int convert_to_spdif_status(unsigned short val) if (val & AC_DIG1_PROFESSIONAL) sbits |= IEC958_AES0_PROFESSIONAL; if (sbits & IEC958_AES0_PROFESSIONAL) { - if (sbits & AC_DIG1_EMPHASIS) + if (val & AC_DIG1_EMPHASIS) sbits |= IEC958_AES0_PRO_EMPHASIS_5015; } else { if (val & AC_DIG1_EMPHASIS) -- cgit v1.1 From eb49faa6a4703698fa5d8b304b01e7f59e7d1f11 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Mar 2013 09:19:11 +0100 Subject: ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader The current DSP loader code abuses snd_hda_lock_devices() for ensuring the DSP loader not conflicting with the other normal operations. But this trick obviously doesn't work for the PM resume since the streams are kept opened there where snd_hda_lock_devices() returns -EBUSY. That means we need another lock mechanism instead of abuse. This patch provides the new lock state to azx_dev. Theoretically it's possible that the DSP loader conflicts with the stream that has been already assigned for another PCM. If it's running, the DSP loader should simply fail. If not -- it's the case for PM resume --, we should assign this stream temporarily to the DSP loader, and take it back to the PCM after finishing DSP loading. If the PCM is operated during the DSP loading, it should get an error, too. Reported-and-tested-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 132 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 109 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4cea6bb6..418bfc0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,8 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; unsigned int irq_pending :1; + unsigned int prepared:1; + unsigned int locked:1; /* * For VIA: * A flag to ensure DMA position is 0 @@ -426,8 +428,25 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct mutex dsp_mutex; +#endif }; +/* DSP lock helpers */ +#ifdef CONFIG_SND_HDA_DSP_LOADER +#define dsp_lock_init(dev) mutex_init(&(dev)->dsp_mutex) +#define dsp_lock(dev) mutex_lock(&(dev)->dsp_mutex) +#define dsp_unlock(dev) mutex_unlock(&(dev)->dsp_mutex) +#define dsp_is_locked(dev) ((dev)->locked) +#else +#define dsp_lock_init(dev) do {} while (0) +#define dsp_lock(dev) do {} while (0) +#define dsp_unlock(dev) do {} while (0) +#define dsp_is_locked(dev) 0 +#endif + /* CORB/RIRB */ struct azx_rb { u32 *buf; /* CORB/RIRB buffer @@ -527,6 +546,10 @@ struct azx { /* card list (for power_save trigger) */ struct list_head list; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct azx_dev saved_azx_dev; +#endif }; #define CREATE_TRACE_POINTS @@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) dev = chip->capture_index_offset; nums = chip->capture_streams; } - for (i = 0; i < nums; i++, dev++) - if (!chip->azx_dev[dev].opened) { - res = &chip->azx_dev[dev]; - if (res->assigned_key == key) - break; + for (i = 0; i < nums; i++, dev++) { + struct azx_dev *azx_dev = &chip->azx_dev[dev]; + dsp_lock(azx_dev); + if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { + res = azx_dev; + if (res->assigned_key == key) { + res->opened = 1; + res->assigned_key = key; + dsp_unlock(azx_dev); + return azx_dev; + } } + dsp_unlock(azx_dev); + } if (res) { + dsp_lock(res); res->opened = 1; res->assigned_key = key; + dsp_unlock(res); } return res; } @@ -2009,6 +2042,12 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct azx_dev *azx_dev = get_azx_dev(substream); int ret; + dsp_lock(azx_dev); + if (dsp_is_locked(azx_dev)) { + ret = -EBUSY; + goto unlock; + } + mark_runtime_wc(chip, azx_dev, substream, false); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; @@ -2016,8 +2055,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto unlock; mark_runtime_wc(chip, azx_dev, substream, true); + unlock: + dsp_unlock(azx_dev); return ret; } @@ -2029,16 +2070,21 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); - azx_sd_writel(azx_dev, SD_CTL, 0); - azx_dev->bufsize = 0; - azx_dev->period_bytes = 0; - azx_dev->format_val = 0; + dsp_lock(azx_dev); + if (!dsp_is_locked(azx_dev)) { + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + } snd_hda_codec_cleanup(apcm->codec, hinfo, substream); mark_runtime_wc(chip, azx_dev, substream, false); + azx_dev->prepared = 0; + dsp_unlock(azx_dev); return snd_pcm_lib_free_pages(substream); } @@ -2055,6 +2101,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid); unsigned short ctls = spdif ? spdif->ctls : 0; + dsp_lock(azx_dev); + if (dsp_is_locked(azx_dev)) { + err = -EBUSY; + goto unlock; + } + azx_stream_reset(chip, azx_dev); format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, @@ -2065,7 +2117,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printk(KERN_ERR SFX "%s: invalid format_val, rate=%d, ch=%d, format=%d\n", pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); - return -EINVAL; + err = -EINVAL; + goto unlock; } bufsize = snd_pcm_lib_buffer_bytes(substream); @@ -2084,7 +2137,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) - return err; + goto unlock; } /* wallclk has 24Mhz clock source */ @@ -2101,8 +2154,14 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) && stream_tag > chip->capture_streams) stream_tag -= chip->capture_streams; - return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, + err = snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, azx_dev->format_val, substream); + + unlock: + if (!err) + azx_dev->prepared = 1; + dsp_unlock(azx_dev); + return err; } static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -2117,6 +2176,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) azx_dev = get_azx_dev(substream); trace_azx_pcm_trigger(chip, azx_dev, cmd); + if (dsp_is_locked(azx_dev) || !azx_dev->prepared) + return -EPIPE; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: rstart = 1; @@ -2621,17 +2683,27 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format, struct azx_dev *azx_dev; int err; - if (snd_hda_lock_devices(bus)) - return -EBUSY; + azx_dev = azx_get_dsp_loader_dev(chip); + + dsp_lock(azx_dev); + spin_lock_irq(&chip->reg_lock); + if (azx_dev->running || azx_dev->locked) { + spin_unlock_irq(&chip->reg_lock); + err = -EBUSY; + goto unlock; + } + azx_dev->prepared = 0; + chip->saved_azx_dev = *azx_dev; + azx_dev->locked = 1; + spin_unlock_irq(&chip->reg_lock); err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), byte_size, bufp); if (err < 0) - goto unlock; + goto err_alloc; mark_pages_wc(chip, bufp, true); - azx_dev = azx_get_dsp_loader_dev(chip); azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; @@ -2649,13 +2721,20 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format, goto error; azx_setup_controller(chip, azx_dev); + dsp_unlock(azx_dev); return azx_dev->stream_tag; error: mark_pages_wc(chip, bufp, false); snd_dma_free_pages(bufp); -unlock: - snd_hda_unlock_devices(bus); + err_alloc: + spin_lock_irq(&chip->reg_lock); + if (azx_dev->opened) + *azx_dev = chip->saved_azx_dev; + azx_dev->locked = 0; + spin_unlock_irq(&chip->reg_lock); + unlock: + dsp_unlock(azx_dev); return err; } @@ -2677,9 +2756,10 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct azx *chip = bus->private_data; struct azx_dev *azx_dev = azx_get_dsp_loader_dev(chip); - if (!dmab->area) + if (!dmab->area || !azx_dev->locked) return; + dsp_lock(azx_dev); /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); @@ -2692,7 +2772,12 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, snd_dma_free_pages(dmab); dmab->area = NULL; - snd_hda_unlock_devices(bus); + spin_lock_irq(&chip->reg_lock); + if (azx_dev->opened) + *azx_dev = chip->saved_azx_dev; + azx_dev->locked = 0; + spin_unlock_irq(&chip->reg_lock); + dsp_unlock(azx_dev); } #endif /* CONFIG_SND_HDA_DSP_LOADER */ @@ -3481,6 +3566,7 @@ static int azx_first_init(struct azx *chip) } for (i = 0; i < chip->num_streams; i++) { + dsp_lock_init(&chip->azx_dev[i]); /* allocate memory for the BDL for each stream */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), -- cgit v1.1 From 55a63d4da3b8850480a1c5b222f77c739e30e346 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Mar 2013 17:20:12 +0100 Subject: ALSA: hda - Fix DAC assignment for independent HP The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 46 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 78897d0..43c2ea5 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -995,6 +995,8 @@ enum { BAD_NO_EXTRA_SURR_DAC = 0x101, /* Primary DAC shared with main surrounds */ BAD_SHARED_SURROUND = 0x100, + /* No independent HP possible */ + BAD_NO_INDEP_HP = 0x40, /* Primary DAC shared with main CLFE */ BAD_SHARED_CLFE = 0x10, /* Primary DAC shared with extra surrounds */ @@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return snd_hda_get_path_idx(codec, path); } +/* check whether the independent HP is available with the current config */ +static bool indep_hp_possible(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct nid_path *path; + int i, idx; + + if (cfg->line_out_type == AUTO_PIN_HP_OUT) + idx = spec->out_paths[0]; + else + idx = spec->hp_paths[0]; + path = snd_hda_get_path_from_idx(codec, idx); + if (!path) + return false; + + /* assume no path conflicts unless aamix is involved */ + if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid)) + return true; + + /* check whether output paths contain aamix */ + for (i = 0; i < cfg->line_outs; i++) { + if (spec->out_paths[i] == idx) + break; + path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]); + if (path && is_nid_contained(path, spec->mixer_nid)) + return false; + } + for (i = 0; i < cfg->speaker_outs; i++) { + path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]); + if (path && is_nid_contained(path, spec->mixer_nid)) + return false; + } + + return true; +} + /* fill the empty entries in the dac array for speaker/hp with the * shared dac pointed by the paths */ @@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec, badness += BAD_MULTI_IO; } + if (spec->indep_hp && !indep_hp_possible(codec)) + badness += BAD_NO_INDEP_HP; + /* re-fill the shared DAC for speaker / headphone */ if (cfg->line_out_type != AUTO_PIN_HP_OUT) refill_shared_dacs(codec, cfg->hp_outs, @@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec) cfg->speaker_pins, val); } + /* clear indep_hp flag if not available */ + if (spec->indep_hp && !indep_hp_possible(codec)) + spec->indep_hp = 0; + kfree(best_cfg); return 0; } -- cgit v1.1