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(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb92732599..4cb49eb8550 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 v1.2.3 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(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4cb49eb8550..a64b93425ae 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 v1.2.3 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(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ab69c83626c..deb109709bf 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 v1.2.3 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(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index a1631320b44..1a97263634e 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 v1.2.3 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(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index fb92fb47d63..1db957d5476 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 v1.2.3 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(-) diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 066f5f3e3f4..c3908862bc8 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 v1.2.3 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(-) diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index c27069d24d7..729958713cd 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 34dc47b9581..a294d942b9f 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 v1.2.3 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(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 857586135d1..0097f3619fa 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 v1.2.3 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(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index deb109709bf..cef288cf999 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 v1.2.3 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(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b57383e8c..3dc656688c0 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 v1.2.3 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(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3dc656688c0..97c68dd24ef 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 v1.2.3 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(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index db02c1e96b0..eefc4563b2f 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 v1.2.3 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(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d4237bc0d8..563c24df4d6 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 v1.2.3 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(+) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 2ffdc35d5ff..806407a3973 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 v1.2.3