From 840d8e5e964dc51673d0f26e119b27d2898e8417 Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Wed, 1 Jun 2011 23:59:10 +0200 Subject: ASoC: atmel_ssc: Don't try to free ssc if request failed We should only call ssc_free() when ssc_request() succeeds or bad things will happen. Signed-off-by: Joachim Eastwood Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 7fbfa051f6e1..eda955b15834 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id) if (IS_ERR(ssc)) pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n", PTR_ERR(ssc)); - else + else { ssc_pdev->dev.parent = &(ssc->pdev->dev); - ssc_free(ssc); + ssc_free(ssc); + } ret = platform_device_add(ssc_pdev); if (ret < 0) -- cgit v1.2.3 From 1622ee1822e8adb391b55a09e3cd5144bd9fad47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2011 17:13:57 +0100 Subject: ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index a0b1a7278284..28fbf072b9c0 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, int old; /* Disable SYSCLK while we reconfigure */ - old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1); + old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA; snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, 0); -- cgit v1.2.3 From 6ac340623c5d2a945030814d900701439772ff57 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2011 18:20:50 +0100 Subject: ASoC: Add missing break in WM8915 FLL source selection Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 28fbf072b9c0..e2ab4fac2819 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, break; case WM8915_FLL_MCLK2: reg = 1; + break; case WM8915_FLL_DACLRCLK1: reg = 2; break; -- cgit v1.2.3 From fd137e2bba53b7207cbae6a1312e89ef3ae55624 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Jun 2011 11:26:15 +0100 Subject: ASoC: Check for NULL register bank in snd_soc_get_cache_val() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81a1601..c005ceb70c9d 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -466,6 +466,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx, static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, unsigned int word_size) { + if (!base) + return -1; + switch (word_size) { case 1: { const u8 *cache = base; -- cgit v1.2.3 From 8ca695f273709a9d147826716a8dee3e0eb2407f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jun 2011 13:38:35 +0200 Subject: ASoC: AD1836: Fix setting the PCM format Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad1836.c | 14 +++++++------- sound/soc/codecs/ad1836.h | 6 ++++++ 2 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ab63d52e36e1..754c496412bd 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; + word_len = AD1836_WORD_LEN_16; break; case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; + word_len = AD1836_WORD_LEN_20; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; + word_len = AD1836_WORD_LEN_24; break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + word_len << AD1836_ADC_WORD_OFFSET); return 0; } diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 845596717fdf..9d6a3f8f8aaf 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -25,6 +25,7 @@ #define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) #define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) #define AD1836_DAC_WORD_LEN_MASK 0x18 +#define AD1836_DAC_WORD_LEN_OFFSET 3 #define AD1836_DAC_CTRL2 1 #define AD1836_DACL1_MUTE 0 @@ -51,6 +52,7 @@ #define AD1836_ADCL2_MUTE 2 #define AD1836_ADCR2_MUTE 3 #define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_WORD_OFFSET 5 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) @@ -60,4 +62,8 @@ #define AD1836_NUM_REGS 16 +#define AD1836_WORD_LEN_24 0x0 +#define AD1836_WORD_LEN_20 0x1 +#define AD1836_WORD_LEN_16 0x2 + #endif -- cgit v1.2.3 From 0a1896b27b030529ec770aefd790544a1bdb7d5a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 6 Jun 2011 18:55:34 -0400 Subject: ALSA: hda: Fix quirk for Dell Inspiron 910 BugLink: https://launchpad.net/bugs/792712 The original reporter states that sound from the internal speakers is inaudible until using the model=auto quirk. This symptom is due to an existing quirk mask for 0x102802b* that uses the model=dell quirk. To limit the possible regressions, leave the existing quirk mask but add a higher priority specific mask for the reporter's PCI SSID. Reported-and-tested-by: rodni hipp Cc: [2.6.38+] Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7a4e10002f56..d7007896772b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13860,6 +13860,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), + SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO), SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), /* almost compatible with toshiba but with optional digital outs; -- cgit v1.2.3 From 064d58ee3afb8a865a72d24e069c7258ec38640e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 7 Jun 2011 10:24:46 +0200 Subject: ASoC: Blackfin: bf5xx-ad1836: Fix codec device name Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/blackfin/bf5xx-ad1836.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index ea4951cf5526..f79d1655e035 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, { @@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, }; -- cgit v1.2.3 From 0f82bdf572fc6e42147151aa4d52542f7fc6d793 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Jun 2011 23:42:04 +0100 Subject: ASoC: Fix WM8962 headphone volume update for use of advanced caches Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f90ae427242b..5e05eed96c38 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, reg_cache[WM8962_HPOUTL_VOLUME]); /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, reg_cache[WM8962_HPOUTR_VOLUME]); -- cgit v1.2.3 From 3115ae174620eeab4b16f52c8d0a9a35d2717e3c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jun 2011 18:07:49 +0100 Subject: ASoC: WM8804 does not support sample rates below 32kHz Reported-by: Kieran O'Leary Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8804.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6785688f8806..9a5e67c5a6bd 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = { #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) +#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + static struct snd_soc_dai_driver wm8804_dai = { .name = "wm8804-spdif", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .ops = &wm8804_dai_ops, -- cgit v1.2.3 From 0cd114fff9ace7014c0d3ef8ab385fc5d3cf2d2f Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Wed, 8 Jun 2011 15:02:56 -0500 Subject: ASoC: fsl: fix initialization of DMA buffers The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs for playback and capture, so DMA buffers should be allocated only for the initialized streams. Instead of checking for the number of active channels, which apparently is not reliable, check to see if the actual stream object exists. Also provide a better name for the DMA interrupt. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 15dac0f20cd8..6680c0b4d203 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, * should allocate a DMA buffer only for the streams that are valid. */ - if (dai->driver->playback.channels_min) { + if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); @@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, } } - if (dai->driver->capture.channels_min) { + if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't alloc capture dma buffer\n"); + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); return ret; } } @@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio", + dma_private); if (ret) { dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); -- cgit v1.2.3 From 4b80b8c2eee5282dab57f094fd3893c0c09f750c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 9 Jun 2011 13:22:36 +0200 Subject: ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a DAPM context not matching the widgets context. This can lead to a wrong prefix_len calculation, which will result in undefined behaviour. To avoid this always use the DAPM context from the widget itself. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 776e6f418306..32ab7fc4579a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, } /* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; int i, ret = 0; size_t name_len, prefix_len; struct snd_soc_dapm_path *path; @@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, } /* create new dapm mux control */ -static int dapm_new_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mux(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; @@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, } /* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_pga(struct snd_soc_dapm_widget *w) { if (w->num_kcontrols) dev_err(w->dapm->dev, @@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: w->power_check = dapm_generic_check_power; - dapm_new_mixer(dapm, w); + dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; - dapm_new_mux(dapm, w); + dapm_new_mux(w); break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: @@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: w->power_check = dapm_generic_check_power; - dapm_new_pga(dapm, w); + dapm_new_pga(w); break; case snd_soc_dapm_input: case snd_soc_dapm_output: -- cgit v1.2.3 From 33195500edf260e8c8809ab9dfc67f50e0ce031f Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Fri, 10 Jun 2011 10:36:54 +0900 Subject: ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register If DMA active status should be checked, I2SCON register should be referenced. In this patch, Fix the incorrect referencing of I2SCON register. Reported-by : Lakkyung Jung Signed-off-by: Sangbeom Kim Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/samsung/i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ffa09b3b2caa..992a732b5211 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD); + active = readl(i2s->addr + I2SCON); if (is_secondary(i2s)) active &= CON_TXSDMA_ACTIVE; @@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE; + active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE; return active ? true : false; } -- cgit v1.2.3 From 20f5e0b36d968326fab3b720035f226113e34ae9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 09:31:54 +0200 Subject: ALSA: hda - Fix invalid unsol tag for some alc262 model quirks The tag number was forgotten to be fixed after cleaning up the model quirks for ALC262 fujitsu and lenovo-3000 models. Tested-by: Michal Hocko Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d7007896772b..ca211c1cba03 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11924,7 +11924,7 @@ static const struct hda_verb alc262_nec_verbs[] = { * 0x1b = port replicator headphone out */ -#define ALC_HP_EVENT 0x37 +#define ALC_HP_EVENT ALC880_HP_EVENT static const struct hda_verb alc262_fujitsu_unsol_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, -- cgit v1.2.3 From c0a20263dbe1fc5f394913d71063c9cd8282c5db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 15:28:15 +0200 Subject: ALSA: hda - Fix initialization of hp pins with master_mute in Realtek Some Reatlek model quirks use master_mute bool switch for controlling the master-mute of outputs. For these cases, the initialization of HP pins/amps were forgotten during the transition to the common automute helper function in 3.0 development time, and resulted in the muted HP output as default. This patch fixes the issue by adjusting the HP output explicitly with master_mute switch. Tested-by: Michal Hocko Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ca211c1cba03..43fcfbd32847 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int on; + /* Control HP pins/amps depending on master_mute state; + * in general, HP pins/amps control should be enabled in all cases, + * but currently set only for master_mute, just to be safe + */ + do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + spec->autocfg.hp_pins, spec->master_mute, true); + if (!spec->automute) on = 0; else @@ -6201,11 +6208,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = { /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; - - /* change HP pins */ - do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), - spec->autocfg.hp_pins, spec->master_mute, true); update_speakers(codec); } -- cgit v1.2.3 From 890ee02ac12c02c4712b6d7dd062ff4d6d37691c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 15:32:31 +0200 Subject: ALSA: Use %pV for snd_printk() Clean up snd_printk() helper using the %pV prefix for recursive printks. This also automagically fixes an Oops with RO/NX-enabled modules. Tested-by: Maarten Lankhorst Signed-off-by: Takashi Iwai --- sound/core/misc.c | 40 +++++++++++++++++----------------------- 1 file changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/core/misc.c b/sound/core/misc.c index 2c41825c836e..eb9fe2e1d291 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -58,26 +58,6 @@ static const char *sanity_file_name(const char *path) else return path; } - -/* print file and line with a certain printk prefix */ -static int print_snd_pfx(unsigned int level, const char *path, int line, - const char *format) -{ - const char *file = sanity_file_name(path); - char tmp[] = "<0>"; - const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT; - int ret = 0; - - if (format[0] == '<' && format[2] == '>') { - tmp[1] = format[1]; - pfx = tmp; - ret = 1; - } - printk("%sALSA %s:%d: ", pfx, file, line); - return ret; -} -#else -#define print_snd_pfx(level, path, line, format) 0 #endif #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) @@ -85,15 +65,29 @@ void __snd_printk(unsigned int level, const char *path, int line, const char *format, ...) { va_list args; - +#ifdef CONFIG_SND_VERBOSE_PRINTK + struct va_format vaf; + char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; +#endif + #ifdef CONFIG_SND_DEBUG if (debug < level) return; #endif + va_start(args, format); - if (print_snd_pfx(level, path, line, format)) - format += 3; /* skip the printk level-prefix */ +#ifdef CONFIG_SND_VERBOSE_PRINTK + vaf.fmt = format; + vaf.va = &args; + if (format[0] == '<' && format[2] == '>') { + memcpy(verbose_fmt, format, 3); + vaf.fmt = format + 3; + } else if (level) + memcpy(verbose_fmt, KERN_DEBUG, 3); + printk(verbose_fmt, sanity_file_name(path), line, &vaf); +#else vprintk(format, args); +#endif va_end(args); } EXPORT_SYMBOL_GPL(__snd_printk); -- cgit v1.2.3 From 7ab1fc0af3464d231e17eb729a03495d93d0cc5c Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 10 Jun 2011 10:14:01 -0400 Subject: ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop BugLink: https://launchpad.net/bugs/761171 The original reporter needs the model=auto quirk for his internal speakers to be audible in the latest daily snapshot, so add an entry in the quirk table for his PCI SSID. A trivially different version of this patch using the model=asus quirk should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much improved. Reported-and-tested-by: tomdeering7 Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3e6b9a8539c2..694b9daf691f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3102,6 +3102,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ + SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; -- cgit v1.2.3