From 8e8b2d676f3f7c1246b108793fb5690e6c6fcd26 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Mon, 12 Oct 2009 08:41:59 +0300 Subject: ASoC: Serialize access to dapm_power_widgets() Access to damp_power_widgets() is assumed to be single-threaded. Concurrent accesses to dapm_power_widgets() may result in unpredictable behavior. Calls from: close_delayed_work() soc_codec_close() soc_pcm_prepare() soc_suspend() soc_resume_deferred() to snd_soc_dapm_stream_event() do not have the codec->mutex taken to cover the call to dapm_power_widgets(). Thus, take the mutex in these paths also to assure single-threaded use of dapm_power_widgets(). Signed-off-by: Eero Nurkkala Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8de6f9dec4a..d89f6dc0090 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2072,9 +2072,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, } } } - mutex_unlock(&codec->mutex); dapm_power_widgets(codec, event); + mutex_unlock(&codec->mutex); dump_dapm(codec, __func__); return 0; } -- cgit v1.2.3-70-g09d2 From 4b7348a15972274eb16182d10987f69da3e95719 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Oct 2009 18:25:23 +0200 Subject: ALSA: hda - Fix capture source checks for ALC662/663 codecs The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls() to check the capture source selections. This should be alc882, instead. Reference: Novell bnc#546918 http://bugzilla.novell.com/show_bug.cgi?id=546918 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c08ca660dab..9b1cff83497 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17374,7 +17374,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, /* create playback/capture controls for input pins */ #define alc662_auto_create_input_ctls \ - alc880_auto_create_input_ctls + alc882_auto_create_input_ctls static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, -- cgit v1.2.3-70-g09d2 From 02a06d3042e208cb74369838b178ca9512192be4 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 16 Oct 2009 18:13:38 +0800 Subject: ASoC: Fix possible codec_dai->ops NULL pointer problems Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc. access the ops field in these DAIs, panic will happen. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad2a97..0a1b2f64bbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device); #define soc_resume NULL #endif +static struct snd_soc_dai_ops null_dai_ops = { +}; + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct platform_device *pdev = container_of(card->dev, @@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ac97 = 1; } + for (i = 0; i < card->num_links; i++) { + if (!card->dai_link[i].codec_dai->ops) + card->dai_link[i].codec_dai->ops = &null_dai_ops; + } + /* If we have AC97 in the system then don't wait for the * codec. This will need revisiting if we have to handle * systems with mixed AC97 and non-AC97 parts. Only check for @@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } -static struct snd_soc_dai_ops null_dai_ops = { -}; - /** * snd_soc_register_dai - Register a DAI with the ASoC core * -- cgit v1.2.3-70-g09d2 From b214f11fb92713fbb07d8c1f62dd1aa8077b56c9 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sat, 24 Oct 2009 00:06:48 +0200 Subject: ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text I thought it could be usefull to add some information on how to get the device fully supported by loading a line discipline on the modem line. Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee9839be8..653a362425d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 help - Say Y if you want to add support for SoC audio on Amstrad Delta. + Say Y if you want to add support for SoC audio device connected to + a handset and a speakerphone found on Amstrad E3 (Delta) videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" -- cgit v1.2.3-70-g09d2 From e3d8024891dbfec6cf36c9b76177650f48118462 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Tue, 22 Sep 2009 16:48:56 +0100 Subject: ARM: S3C: Add info for supporting circular DMA buffers The S3C64XX DMA implementation will work a lot better with the ability to enqueue circular buffers as the hardware can do it's own linked-list management. Add a function s3c_dma_has_circular() to show that the system can do this and a flag for the channel. Update the s3c24xx/s3c64xx I2S DMA code to deal with this. Signed-off-by: Ben Dooks Signed-off-by: Ben Dooks Acked-by: Mark Brown --- arch/arm/mach-s3c2410/include/mach/dma.h | 7 +++++++ arch/arm/mach-s3c6400/include/mach/dma.h | 5 +++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 17 +++++++++++++++-- 3 files changed, 27 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-s3c2410/include/mach/dma.h b/arch/arm/mach-s3c2410/include/mach/dma.h index c3a2629e0de..92e2687009e 100644 --- a/arch/arm/mach-s3c2410/include/mach/dma.h +++ b/arch/arm/mach-s3c2410/include/mach/dma.h @@ -110,6 +110,8 @@ enum s3c2410_dma_loadst { * waiting for reloads */ #define S3C2410_DMAF_AUTOSTART (1<<1) /* auto-start if buffer queued */ +#define S3C2410_DMAF_CIRCULAR (1 << 2) /* no circular dma support */ + /* dma buffer */ struct s3c2410_dma_buf; @@ -194,4 +196,9 @@ struct s3c2410_dma_chan { typedef unsigned long dma_device_t; +static inline bool s3c_dma_has_circular(void) +{ + return false; +} + #endif /* __ASM_ARCH_DMA_H */ diff --git a/arch/arm/mach-s3c6400/include/mach/dma.h b/arch/arm/mach-s3c6400/include/mach/dma.h index 1067619f0ba..004edab2395 100644 --- a/arch/arm/mach-s3c6400/include/mach/dma.h +++ b/arch/arm/mach-s3c6400/include/mach/dma.h @@ -68,6 +68,11 @@ static __inline__ int s3c_dma_has_circular(void) #define S3C2410_DMAF_CIRCULAR (1 << 0) +static inline bool s3c_dma_has_circular(void) +{ + return false; +} + #include #endif /* __ASM_ARCH_IRQ_H */ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 5cbbdc80fde..1f35c6fcf5f 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; + unsigned int limit; int ret; pr_debug("Entered %s\n", __func__); - while (prtd->dma_loaded < prtd->dma_limit) { + if (s3c_dma_has_circular()) { + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + } else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { unsigned long len = prtd->dma_period; pr_debug("dma_loaded: %d\n", prtd->dma_loaded); @@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING) { + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; s3c24xx_pcm_enqueue(substream); } @@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, printk(KERN_ERR "failed to get dma channel\n"); return ret; } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); } s3c2410_dma_set_buffdone_fn(prtd->params->channel, -- cgit v1.2.3-70-g09d2 From 97609458ce972180172ae2cec0483451820e6a41 Mon Sep 17 00:00:00 2001 From: Wu Zhangjin Date: Thu, 15 Oct 2009 10:22:54 +0800 Subject: ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency SND_CS5535AUDIO is available on Loongson(MIPS compatible) family machines, and checked it with ARCH=x86_64, no relative compiling warnings & errors, so, remove the platform dependency directly. Reported-by: rixed@happyleptic.org Acked-by: Andres Salomon Signed-off-by: Wu Zhangjin Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fb5ee3cc396..75c602b5b13 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -259,7 +259,6 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help -- cgit v1.2.3-70-g09d2 From b71207e9dc044b30d8b5d7f1c2290ba14563f05c Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Fri, 30 Oct 2009 11:51:24 +0100 Subject: ALSA: pcsp - Fix nforce workaround The attached patch fixes the problems introduced in this commit: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=eea0579fc85e64e9f05361d5aacf496fe7a151aa - Fix nForce workaround by honouring the pointer_update var - Revert "ns" to u64, as per the hrtimer API - Revert to the zero-delay timer startup, since I can't reproduce any problem with it (please, give me the hint!) Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_lib.c | 65 +++++++++++++++++++++-------------------- sound/drivers/pcsp/pcsp_mixer.c | 2 +- 2 files changed, 35 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 84cc2658c05..e1145ac6e90 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -39,25 +39,20 @@ static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0); /* write the port and returns the next expire time in ns; * called at the trigger-start and in hrtimer callback */ -static unsigned long pcsp_timer_update(struct hrtimer *handle) +static u64 pcsp_timer_update(struct snd_pcsp *chip) { unsigned char timer_cnt, val; u64 ns; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; - struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); unsigned long flags; if (chip->thalf) { outb(chip->val61, 0x61); chip->thalf = 0; - if (!atomic_read(&chip->timer_active)) - return 0; return chip->ns_rem; } - if (!atomic_read(&chip->timer_active)) - return 0; substream = chip->playback_substream; if (!substream) return 0; @@ -88,24 +83,17 @@ static unsigned long pcsp_timer_update(struct hrtimer *handle) return ns; } -enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +static void pcsp_pointer_update(struct snd_pcsp *chip) { - struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); struct snd_pcm_substream *substream; - int periods_elapsed, pointer_update; size_t period_bytes, buffer_bytes; - unsigned long ns; + int periods_elapsed; unsigned long flags; - pointer_update = !chip->thalf; - ns = pcsp_timer_update(handle); - if (!ns) - return HRTIMER_NORESTART; - /* update the playback position */ substream = chip->playback_substream; if (!substream) - return HRTIMER_NORESTART; + return; period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); @@ -134,6 +122,26 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (periods_elapsed) tasklet_schedule(&pcsp_pcm_tasklet); +} + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + int pointer_update; + u64 ns; + + if (!atomic_read(&chip->timer_active) || !chip->playback_substream) + return HRTIMER_NORESTART; + + pointer_update = !chip->thalf; + ns = pcsp_timer_update(chip); + if (!ns) { + printk(KERN_WARNING "PCSP: unexpected stop\n"); + return HRTIMER_NORESTART; + } + + if (pointer_update) + pcsp_pointer_update(chip); hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); @@ -142,8 +150,6 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) static int pcsp_start_playing(struct snd_pcsp *chip) { - unsigned long ns; - #if PCSP_DEBUG printk(KERN_INFO "PCSP: start_playing called\n"); #endif @@ -159,11 +165,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip) atomic_set(&chip->timer_active, 1); chip->thalf = 0; - ns = pcsp_timer_update(&pcsp_chip.timer); - if (!ns) - return -EIO; - - hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL); + hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); return 0; } @@ -232,21 +234,22 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); + pcsp_sync_stop(chip); + chip->playback_ptr = 0; + chip->period_ptr = 0; + chip->fmt_size = + snd_pcm_format_physical_width(substream->runtime->format) >> 3; + chip->is_signed = snd_pcm_format_signed(substream->runtime->format); #if PCSP_DEBUG printk(KERN_INFO "PCSP: prepare called, " - "size=%zi psize=%zi f=%zi f1=%i\n", + "size=%zi psize=%zi f=%zi f1=%i fsize=%i\n", snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream) / snd_pcm_lib_period_bytes(substream), - substream->runtime->periods); + substream->runtime->periods, + chip->fmt_size); #endif - pcsp_sync_stop(chip); - chip->playback_ptr = 0; - chip->period_ptr = 0; - chip->fmt_size = - snd_pcm_format_physical_width(substream->runtime->format) >> 3; - chip->is_signed = snd_pcm_format_signed(substream->runtime->format); return 0; } diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 199b0337714..903bc846763 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -72,7 +72,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol, if (treble != chip->treble) { chip->treble = treble; #if PCSP_DEBUG - printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE()); + printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE()); #endif changed = 1; } -- cgit v1.2.3-70-g09d2 From db32f99816f7cbe61c1f75c1560655a3bf52488a Mon Sep 17 00:00:00 2001 From: peer chen Date: Thu, 15 Oct 2009 16:37:47 +0800 Subject: ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller Add the generic device ID for NVIDIA HDA controller. Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c9ad182e1b4..e340792f6cb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2674,6 +2674,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, -- cgit v1.2.3-70-g09d2 From 4b3be6afa4ab8b3fdce39df68bad71f8b85164de Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 17 Oct 2009 08:33:22 +0200 Subject: ALSA: sound: Move dereference after NULL test and drop unnecessary NULL tests In pcm.c, if the NULL test on pcm is needed, then the dereference should be after the NULL test. In dummy.c and ali5451.c, the context of the calls to snd_card_dummy_new_mixer and snd_ali_free_voice show that dummy and pvoice, respectively cannot be NULL. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 5 +++-- sound/drivers/dummy.c | 2 -- sound/pci/ali5451/ali5451.c | 2 +- 3 files changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 0c1440121c2..c69c60b2a48 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -953,11 +953,12 @@ static int snd_pcm_dev_register(struct snd_device *device) struct snd_pcm_substream *substream; struct snd_pcm_notify *notify; char str[16]; - struct snd_pcm *pcm = device->device_data; + struct snd_pcm *pcm; struct device *dev; - if (snd_BUG_ON(!pcm || !device)) + if (snd_BUG_ON(!device || !device->device_data)) return -ENXIO; + pcm = device->device_data; mutex_lock(®ister_mutex); err = snd_pcm_add(pcm); if (err) { diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 6ba066c41d2..146ef00f94a 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -808,8 +808,6 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) unsigned int idx; int err; - if (snd_BUG_ON(!dummy)) - return -EINVAL; spin_lock_init(&dummy->mixer_lock); strcpy(card->mixername, "Dummy Mixer"); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index b458d208720..aaf4da68969 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec, void *private_data; snd_ali_printk("free_voice: channel=%d\n",pvoice->number); - if (pvoice == NULL || !pvoice->use) + if (!pvoice->use) return; snd_ali_clear_voices(codec, pvoice->number, pvoice->number); spin_lock_irq(&codec->voice_alloc); -- cgit v1.2.3-70-g09d2 From e8e0929d7290cab7c5b1a3e5f5f54f73daf38038 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 17 Oct 2009 08:33:47 +0200 Subject: ALSA: sound/parisc: Move dereference after NULL test If the NULL test on h is needed in snd_harmony_mixer_init, then the dereference should be after the NULL test. Actually, there is a sequence of calls: snd_harmony_create, then snd_harmony_pcm_init, and then snd_harmony_mixer_init. snd_harmony_create initializes h, but may indeed leave it as NULL. There was no NULL test at the beginning of snd_harmony_pcm_init, so I have added one. The NULL test in snd_harmony_mixer_init is then not necessary, but in case the ordering of the calls changes, I have left it, and moved the dereference after it. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/parisc/harmony.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index e924492df21..f47f9e226b0 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -624,6 +624,9 @@ snd_harmony_pcm_init(struct snd_harmony *h) struct snd_pcm *pcm; int err; + if (snd_BUG_ON(!h)) + return -EINVAL; + harmony_disable_interrupts(h); err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm); @@ -865,11 +868,12 @@ snd_harmony_mixer_reset(struct snd_harmony *h) static int __devinit snd_harmony_mixer_init(struct snd_harmony *h) { - struct snd_card *card = h->card; + struct snd_card *card; int idx, err; if (snd_BUG_ON(!h)) return -EINVAL; + card = h->card; strcpy(card->mixername, "Harmony Gain control interface"); for (idx = 0; idx < HARMONY_CONTROLS; idx++) { -- cgit v1.2.3-70-g09d2 From 3702b082281929cf1bdf14f67eb0619aab58b496 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Sat, 24 Oct 2009 12:59:35 +0100 Subject: ALSA: snd-usb-caiaq: Missing lock around use of buffer positions Fix a race which causes snd_pcm_update_hw_ptr_pos() to report a bug. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 121af0644fd..e76017cd5ac 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -269,16 +269,22 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) { int index = sub->number; struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + snd_pcm_uframes_t ptr; + + spin_lock(&dev->spinlock); if (dev->input_panic || dev->output_panic) - return SNDRV_PCM_POS_XRUN; + ptr = SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); + + spin_unlock(&dev->spinlock); + return ptr; } /* operators for both playback and capture */ -- cgit v1.2.3-70-g09d2 From ac9dd9d384b018f1e1c5a9a2686ab5605ce55818 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Sat, 24 Oct 2009 12:59:36 +0100 Subject: ALSA: snd-usb-caiaq: Lock on stream start/unpause Fix a bug which can result in white noise from the driver after stream start or unpause. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index e76017cd5ac..86b2c3b92df 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -62,10 +62,14 @@ static void activate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { + spin_lock(&dev->spinlock); + if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->sub_playback[sub->number] = sub; else dev->sub_capture[sub->number] = sub; + + spin_unlock(&dev->spinlock); } static void -- cgit v1.2.3-70-g09d2 From 467cc1692036909ee0a723ce633fc4a53d72fd9a Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Sat, 24 Oct 2009 12:59:37 +0100 Subject: ALSA: snd-usb-caiaq: Bump version number to 1.3.20 Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 83e6c1312d4..a3f02dd9744 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- cgit v1.2.3-70-g09d2 From 3d00941371a765779c4e3509214c7e5793cce1fe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 22 Oct 2009 09:04:09 +0200 Subject: sound: via82xx: deactivate DXS controls of inactive streams Activate the DXS volume controls only when the corresponding stream is being used. This makes the behaviour consistent with the other drivers that have per-stream volume controls. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 59 ++++++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 52 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 91683a34903..8a332d2f615 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -386,6 +386,7 @@ struct via82xx { struct snd_pcm *pcms[2]; struct snd_rawmidi *rmidi; + struct snd_kcontrol *dxs_controls[4]; struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97; @@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, /* - * open callback for playback on via686 and via823x DSX + * open callback for playback on via686 */ -static int snd_via82xx_playback_open(struct snd_pcm_substream *substream) +static int snd_via686_playback_open(struct snd_pcm_substream *substream) { struct via82xx *chip = snd_pcm_substream_chip(substream); struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number]; @@ -1229,6 +1230,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream) return 0; } +/* + * open callback for playback on via823x DXS + */ +static int snd_via8233_playback_open(struct snd_pcm_substream *substream) +{ + struct via82xx *chip = snd_pcm_substream_chip(substream); + struct viadev *viadev; + unsigned int stream; + int err; + + viadev = &chip->devs[chip->playback_devno + substream->number]; + if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0) + return err; + stream = viadev->reg_offset / 0x10; + if (chip->dxs_controls[stream]) { + chip->playback_volume[stream][0] = 0; + chip->playback_volume[stream][1] = 0; + chip->dxs_controls[stream]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &chip->dxs_controls[stream]->id); + } + return 0; +} + /* * open callback for playback on via823x multi-channel */ @@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream) return 0; } +static int snd_via8233_playback_close(struct snd_pcm_substream *substream) +{ + struct via82xx *chip = snd_pcm_substream_chip(substream); + struct viadev *viadev = substream->runtime->private_data; + unsigned int stream; + + stream = viadev->reg_offset / 0x10; + if (chip->dxs_controls[stream]) { + chip->dxs_controls[stream]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &chip->dxs_controls[stream]->id); + } + return snd_via82xx_pcm_close(substream); +} + /* via686 playback callbacks */ static struct snd_pcm_ops snd_via686_playback_ops = { - .open = snd_via82xx_playback_open, + .open = snd_via686_playback_open, .close = snd_via82xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_via82xx_hw_params, @@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = { /* via823x DSX playback callbacks */ static struct snd_pcm_ops snd_via8233_playback_ops = { - .open = snd_via82xx_playback_open, - .close = snd_via82xx_pcm_close, + .open = snd_via8233_playback_open, + .close = snd_via8233_playback_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_via82xx_hw_params, .hw_free = snd_via82xx_hw_free, @@ -1709,8 +1752,9 @@ static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { .device = 0, /* .subdevice set later */ .name = "PCM Playback Volume", - .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, .info = snd_via8233_dxs_volume_info, .get = snd_via8233_dxs_volume_get, .put = snd_via8233_dxs_volume_put, @@ -1948,6 +1992,7 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) err = snd_ctl_add(chip->card, kctl); if (err < 0) return err; + chip->dxs_controls[i] = kctl; } } } -- cgit v1.2.3-70-g09d2 From a1bf808849f25a4d668f81415ecebb2da9fecf8e Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 1 Nov 2009 18:32:29 -0500 Subject: ALSA: hda: Use quirk mask for Dell Inspiron Mini9/Vostro A90 using ALC268 BugLink: https://bugs.launchpad.net/bugs/368629 We should use a quirk mask for these Dell Inspiron Mini9s and Vostro A90s, as the model=dell quirk appears to enable audio on them. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b1cff83497..148734d1613 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12602,7 +12602,8 @@ static 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 Mini9", ALC268_DELL), + SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, + "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), /* almost compatible with toshiba but with optional digital outs; * auto-probing seems working fine */ -- cgit v1.2.3-70-g09d2 From 0d488234fd857aae07f1c56467bbf58f1a859753 Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Sat, 24 Oct 2009 21:43:03 +0200 Subject: ALSA: pcmcia: use dynamic debug infrastructure, deprecate CS_CHECK (sound) Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of requiring manual settings of PCMCIA_DEBUG. Also, remove all usages of the CS_CHECK macro and replace them with proper Linux style calling and return value checking. The extra error reporting may be dropped, as the PCMCIA core already complains about any (non-driver-author) errors. Signed-off-by: Dominik Brodowski Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 21 ++++++++++++--------- sound/pcmcia/vx/vxpocket.c | 21 ++++++++++++--------- 2 files changed, 24 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7dea74b71cf..64b859925c0 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -217,20 +217,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link) * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int pdacf_config(struct pcmcia_device *link) { struct snd_pdacf *pdacf = link->priv; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->conf.ConfigIndex = 0x5; - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0) goto failed; @@ -238,8 +243,6 @@ static int pdacf_config(struct pcmcia_device *link) link->dev_node = &pdacf->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7445cc8a47d..1492744ad67 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -213,14 +213,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int vxpocket_config(struct pcmcia_device *link) { struct vx_core *chip = link->priv; struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "vxpocket_config called\n"); @@ -235,9 +232,17 @@ static int vxpocket_config(struct pcmcia_device *link) strcpy(chip->card->driver, vxp440_hw.name); } - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; chip->dev = &handle_to_dev(link); snd_card_set_dev(chip->card, chip->dev); @@ -248,8 +253,6 @@ static int vxpocket_config(struct pcmcia_device *link) link->dev_node = &vxp->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; -- cgit v1.2.3-70-g09d2 From 23aebca486429b74c35b41ac5cac7ce97609fd6a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Nov 2009 14:10:59 +0100 Subject: ALSA: dummy - Fix descriptions of pcm_substreams parameter Now up to 128 substreams are supported. Reported-by: Adrian Bridgett Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/drivers/dummy.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb4518ce..fd9a2f67edf 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -522,7 +522,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. pcm_devs - Number of PCM devices assigned to each card (default = 1, up to 4) pcm_substreams - Number of PCM substreams assigned to each PCM - (default = 8, up to 16) + (default = 8, up to 128) hrtimer - Use hrtimer (=1, default) or system timer (=0) fake_buffer - Fake buffer allocations (default = 1) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 146ef00f94a..252e04ce602 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -165,7 +165,7 @@ MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); -MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver."); +MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); module_param(fake_buffer, bool, 0444); -- cgit v1.2.3-70-g09d2 From ad87c64f00e01a694bf90bddc2b4a6c90796d13c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Nov 2009 14:23:15 +0100 Subject: ALSA: hda - Don't check invalid HP pin alc_automute_pin() might be called even if any HP pin is defined, and it will result in verbs with NID=0. This patch adds a check for the validity of HP widget before issuing any verbs. 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 148734d1613..ff20048504b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -965,6 +965,8 @@ static void alc_automute_pin(struct hda_codec *codec) unsigned int nid = spec->autocfg.hp_pins[0]; int i; + if (!nid) + return; pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); -- cgit v1.2.3-70-g09d2 From 5bdaaada16363d64e10ae081755d1a8d392429f2 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Wed, 4 Nov 2009 07:57:45 +0100 Subject: ALSA: hda - Enable GPIO control for mute LED on HP systems This patch enables GPIO to control mute LED indicator on the HP systems with the special string in BIOS and applies it with the correct polarity on HP B-series systems. It also restores configuration of the pin intended as the second Headphone on HP B-series systems but configured as something else in the BIOS to pass MS DTM. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 68 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 68 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 66c0876bf73..b513eba2d2f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -1693,6 +1694,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, "HP dv4-1222nr", STAC_HP_DV4_1222NR), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720, + "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -4730,6 +4733,26 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } +static int hp_bseries_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + return 1; + } + return 0; +} + /* * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) @@ -4754,6 +4777,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, else spec->gpio_data |= spec->gpio_led; /* white */ + if (hp_bseries_system(codec->subsystem_id)) { + /* LED state is inverted on these systems */ + spec->gpio_data ^= spec->gpio_led; + } + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); @@ -5243,6 +5271,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; + unsigned int pin_cfg; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5426,6 +5455,45 @@ again: break; } + if (hp_bseries_system(codec->subsystem_id)) { + pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); + if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || + get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || + get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) { + /* It was changed in the BIOS to just satisfy MS DTM. + * Lets turn it back into slaved HP + */ + pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) + | (AC_JACK_HP_OUT << + AC_DEFCFG_DEVICE_SHIFT); + pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC + | AC_DEFCFG_SEQUENCE))) + | 0x1f; + snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg); + } + } + + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + const struct dmi_device *dev = NULL; + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, + NULL, dev))) { + if (strcmp(dev->name, "HP_Mute_LED_1")) { + switch (codec->vendor_id) { + case 0x111d7608: + spec->gpio_led = 0x01; + break; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + spec->gpio_led = 0x08; + break; + } + break; + } + } + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { spec->gpio_mask |= spec->gpio_led; -- cgit v1.2.3-70-g09d2 From 798a8a15011e88cc63dbbb15728b42572c152093 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Wed, 4 Nov 2009 10:11:07 +0000 Subject: ALSA: hda - Add OLPC XO-1.5 PCI ID The XO-1.5 laptop now has a unique subvendor/subproduct ID, which can be used to automatically select the correct CXT5066 configuration. Signed-off-by: Daniel Drake Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3fbbc8c01e7..6479e65858d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2325,6 +2325,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_LAPTOP), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), {} }; -- cgit v1.2.3-70-g09d2 From 7e6c3989af5baee999ef9a4424e85938cba8d34a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 4 Nov 2009 21:03:46 -0500 Subject: ALSA: intel8x0: Mute External Amplifier by default for another Sony model BugLink: https://bugs.launchpad.net/bugs/474972 This Sony model needs External Amplifier muted for audible playback, so make sure we set the inv_eapd quirk. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 754867ed478..aac20fb4aad 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1948,6 +1948,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP xw4200", /* AD1981B*/ .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x104d, + .subdevice = 0x8144, + .name = "Sony", + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x104d, .subdevice = 0x8197, -- cgit v1.2.3-70-g09d2 From f702cf463e1308fbb0c1faa9f3d8e3fa9cb5630f Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 4 Nov 2009 16:04:52 -0800 Subject: sound: Use KERN_WARNING instead of KERN_WARN, which does not exist Reported-by: Andrew Lyon Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/oss/sb_common.c | 4 ++-- sound/oss/sb_ess.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 77d0e5efda7..ce4db49291f 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -157,7 +157,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } @@ -177,7 +177,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 180e95c87e3..51a3d381a59 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -782,7 +782,7 @@ printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode); break; default:; - /* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */ } } -- cgit v1.2.3-70-g09d2 From 78987bdc4e41a425ac113c2c51474f0368fe653a Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Thu, 5 Nov 2009 09:22:30 -0800 Subject: ALSA: hda, move hp_bseries_system Function hp_bseries_system() is always used, outside of CONFIG_ boundaries/controls, so move it. sound/pci/hda/patch_sigmatel.c:5458: error: implicit declaration of function 'hp_bseries_system' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b513eba2d2f..8eb6508cd99 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4668,6 +4668,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_bseries_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + return 1; + } + return 0; +} + #ifdef CONFIG_PROC_FS static void stac92hd_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) @@ -4733,26 +4753,6 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } -static int hp_bseries_system(u32 subsystem_id) -{ - switch (subsystem_id) { - case 0x103c307e: - case 0x103c307f: - case 0x103c3080: - case 0x103c3081: - case 0x103c1722: - case 0x103c1723: - case 0x103c1724: - case 0x103c1725: - case 0x103c1726: - case 0x103c1727: - case 0x103c1728: - case 0x103c1729: - return 1; - } - return 0; -} - /* * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) -- cgit v1.2.3-70-g09d2 From 4d187fb830a7aa8afb471124abe41b04caf49401 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 21 Oct 2009 23:10:03 +0200 Subject: ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1 After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c, omap_pcm_prepare() unconditionally calls: omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); Current implementation of those two functions found in arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at all, so they both end with BUG() on that machine. That results in ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta. The patch corrects the problem by not calling those two functions when run on OMAP1 class based machines. Created against linux-2.6.32-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5735945788b..6a829eef2a4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) else omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); - omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); - omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); + if (!(cpu_class_is_omap1())) { + omap_set_dma_src_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + omap_set_dma_dest_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + } return 0; } -- cgit v1.2.3-70-g09d2 From 6fc786d5034ed7ce2d43c459211137de6d99dd28 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Fri, 6 Nov 2009 18:00:24 +0900 Subject: ASoC: S3C64XX I2S: Enable audio-bus clock Added the missing clk_enable after acquiring the 'audio-bus' clock. Signed-off-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 3c06c401d0f..105a77eeded 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) goto err; } + clk_enable(i2s->iis_cclk); + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); if (ret) goto err_clk; -- cgit v1.2.3-70-g09d2 From 70edc800a39327174d3244f9226ce8cacd01dc91 Mon Sep 17 00:00:00 2001 From: Thomas Gleixner Date: Fri, 6 Nov 2009 22:41:29 +0000 Subject: sound: Replace old style lock initializer SPIN_LOCK_UNLOCKED is deprecated. Use __SPIN_LOCK_UNLOCKED instead. Signed-off-by: Thomas Gleixner Signed-off-by: Takashi Iwai --- sound/oss/dmasound/dmasound_core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 793b7f47843..3f3c3f71db4 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -219,7 +219,9 @@ static int shared_resources_initialised; * Mid level stuff */ -struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED }; +struct sound_settings dmasound = { + .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock) +}; static inline void sound_silence(void) { -- cgit v1.2.3-70-g09d2 From f495088210c8b9e20791d995a8210170c68d2deb Mon Sep 17 00:00:00 2001 From: Julian Anastasov Date: Fri, 6 Nov 2009 23:44:53 +0200 Subject: ALSA: usb-audio: fix combine_word problem Fix combine_word problem where first octet is not read properly. The only affected place seems to be the INPUT_TERMINAL type. Before now, sound controls can be created with the output terminal's name which is a fallback mechanism used only for unknown input terminal types. For example, Line can wrongly appear as Speaker. After the change it should appear as Line. The side effect of this change can be that users can expect the wrong control name in their scripts or programs while now we return the correct one. Probably, these defines should use get_unaligned_le16 and friends. Signed-off-by: Julian Anastasov Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f78941ba..e9a3a9dca15 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -210,7 +210,7 @@ struct snd_usb_midi_endpoint_info { /* */ -#define combine_word(s) ((*s) | ((unsigned int)(s)[1] << 8)) +#define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) -- cgit v1.2.3-70-g09d2 From 1a6969788ef2d5bc3169eee59def6b267182f136 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 7 Nov 2009 09:49:04 +0100 Subject: ALSA: hda - Don't initialize CORB/RIRB for single_cmd mode So far, CORB/RIRB still remains even if the driver is switched to the single_cmd mode. The specification says that this should be disabled, but I hoped this isn't the case; indeed most devices worked together with CORB/RIRB. However, Poulsbo (US15W) seems problematic with this setup, and it requires to disable CORB/RIRB when single_cmd is used. Now this patch disables CORB/RIRB initialization when the single_cmd mode is used. Also the unsolicited event is disabled because it can't work without RIRB. Reported-and-tested-by: Troy Kisky Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e340792f6cb..6517f589d01 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -722,9 +722,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, chip->last_cmd[addr]); chip->single_cmd = 1; bus->response_reset = 0; - /* re-initialize CORB/RIRB */ + /* release CORB/RIRB */ azx_free_cmd_io(chip); - azx_init_cmd_io(chip); + /* disable unsolicited responses */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL); return -1; } @@ -865,7 +866,9 @@ static int azx_reset(struct azx *chip) } /* Accept unsolicited responses */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL); + if (!chip->single_cmd) + azx_writel(chip, GCTL, azx_readl(chip, GCTL) | + ICH6_GCTL_UNSOL); /* detect codecs */ if (!chip->codec_mask) { @@ -980,7 +983,8 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - azx_init_cmd_io(chip); + if (!chip->single_cmd) + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); -- cgit v1.2.3-70-g09d2 From f37325a956f0ee4356793da7d93c699a25b21d92 Mon Sep 17 00:00:00 2001 From: Ben Hutchings Date: Sat, 7 Nov 2009 22:13:39 +0000 Subject: ALSA: snd-aica: declare MODULE_FIRMWARE Signed-off-by: Ben Hutchings Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 583a3693df7..a0df401ebb9 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -49,6 +49,7 @@ MODULE_AUTHOR("Adrian McMenamin "); MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); +MODULE_FIRMWARE("aica_firmware.bin"); /* module parameters */ #define CARD_NAME "AICA" -- cgit v1.2.3-70-g09d2 From 95491d902b4ed1bfd8f602aada793d74cc85428b Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 8 Nov 2009 19:03:55 -0500 Subject: ALSA: hda: Use model=auto quirk for Sony VAIO VGN-FW170J using ALC262 BugLink: https://bugs.launchpad.net/bugs/478309 The internal microphone on this VAIO model does not work unless the "auto" quirk is used. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ff20048504b..9bb4f75ca43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11460,6 +11460,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), + SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", -- cgit v1.2.3-70-g09d2 From dbaccc0cca830efe9bb3c9e4a1cfcd6503790079 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Mon, 9 Nov 2009 15:17:24 +0000 Subject: ALSA: hda - Tweak OLPC XO-1.5 microphone bias Our contacts at Conexant suggested that we reduce the external microphone bias to 50% in order to center the input signal with the DC input range of the codec. This is because the microphone port is DC coupled for potential use with sensors. Signed-off-by: Daniel Drake Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6479e65858d..905859d4f4d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -110,6 +110,7 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; + unsigned char ext_mic_bias; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; #define CXT5066_SPDIF_OUT 0x21 +/* OLPC's microphone port is DC coupled for use with external sensors, + * therefore we use a 50% mic bias in order to center the input signal with + * the DC input range of the codec. */ +#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50 + static struct hda_channel_mode cxt5066_modes[1] = { { 2, NULL }, }; @@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5066_automic(struct hda_codec *codec) { - static struct hda_verb ext_mic_present[] = { + struct conexant_spec *spec = codec->spec; + struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, /* Port C: internal microphone */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -2353,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; + spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2384,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- cgit v1.2.3-70-g09d2 From 4ac55982907e1d48e64feaa56be91b9b52d3714d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Nov 2009 16:08:45 +0100 Subject: ALSA: hda - Avoid quirk for HP dc5750 The present quirk for HP dc5750 seems broken and maps the pins wrongly. Since the auto-parser works well for this device, set the default entry to use model=auto. Reference: Novell bnc#552154 https://bugzilla.novell.com/show_bug.cgi?id=552154 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bb4f75ca43..d1ccb6eaf9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6249,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), + SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */ SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600), -- cgit v1.2.3-70-g09d2 From 8579d2d7779d7ff41ea2a0183015e0e5038f1043 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:09:38 +0200 Subject: sound: rawmidi: fix double init when opening MIDI device with O_APPEND Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 moved the substream initialization code to where it would be executed every time the substream is opened. This had the consequence that any further opening would drop and leak the data in the existing buffer, and that the device driver's open callback would be called multiple times, unexpectedly. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c0adc14c91f..3071e6f5801 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -266,17 +266,19 @@ static int open_substream(struct snd_rawmidi *rmidi, { int err; - err = snd_rawmidi_runtime_create(substream); - if (err < 0) - return err; - err = substream->ops->open(substream); - if (err < 0) - return err; - substream->opened = 1; - if (substream->use_count++ == 0) + if (substream->use_count == 0) { + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) + return err; + substream->opened = 1; substream->active_sensing = 0; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - substream->append = 1; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + } + substream->use_count++; rmidi->streams[substream->stream].substream_opened++; return 0; } -- cgit v1.2.3-70-g09d2 From 16fb109644b5644e42ececeff644514de6f4bd03 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:10:16 +0200 Subject: sound: rawmidi: fix checking of O_APPEND when opening MIDI device Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 dropped the check that a substream must already have been opened with O_APPEND to be able to open it a second time. This would make it possible for a substream to be switched to append mode, which would mean that non-atomic writes would fail unexpectedly. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 3071e6f5801..091405385e1 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -248,7 +248,8 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { if (stream == SNDRV_RAWMIDI_STREAM_INPUT || - !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + !(mode & SNDRV_RAWMIDI_LFLG_APPEND) || + !substream->append) continue; } if (subdevice < 0 || subdevice == substream->number) { -- cgit v1.2.3-70-g09d2 From b7fe750fcceda4fa6bef399b0e2812562728ea82 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:11:43 +0200 Subject: sound: rawmidi: fix MIDI device O_APPEND error handling Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 broke the error handling code in rawmidi_open_priv(). If only the output substream of a RawMIDI device has been opened and if this device is then opened with O_RDWR | O_APPEND and if the initialization of the input substream fails (either because of low memory or because the device driver's open callback fails), then the runtime structure of the already open output substream will be freed and all following writes through the first handle will cause snd_rawmidi_write() to use the NULL runtime pointer. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 091405385e1..70d6f25ba52 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -272,8 +272,10 @@ static int open_substream(struct snd_rawmidi *rmidi, if (err < 0) return err; err = substream->ops->open(substream); - if (err < 0) + if (err < 0) { + snd_rawmidi_runtime_free(substream); return err; + } substream->opened = 1; substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) @@ -300,27 +302,27 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, SNDRV_RAWMIDI_STREAM_INPUT, mode, &sinput); if (err < 0) - goto __error; + return err; } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { err = assign_substream(rmidi, subdevice, SNDRV_RAWMIDI_STREAM_OUTPUT, mode, &soutput); if (err < 0) - goto __error; + return err; } if (sinput) { err = open_substream(rmidi, sinput, mode); if (err < 0) - goto __error; + return err; } if (soutput) { err = open_substream(rmidi, soutput, mode); if (err < 0) { if (sinput) close_substream(rmidi, sinput, 0); - goto __error; + return err; } } @@ -328,13 +330,6 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, rfile->input = sinput; rfile->output = soutput; return 0; - - __error: - if (sinput && sinput->runtime) - snd_rawmidi_runtime_free(sinput); - if (soutput && soutput->runtime) - snd_rawmidi_runtime_free(soutput); - return err; } /* called from sound/core/seq/seq_midi.c */ -- cgit v1.2.3-70-g09d2 From 71121d9fcc494453b9311992de220abb47dde3f1 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Tue, 10 Nov 2009 20:11:55 +0100 Subject: ALSA: hda - possible read past array alc88[02]_parse_auto_config() The test of index `i' is after the read - too late - and unsafe: if snd_hda_get_connections() fails in the last iteration a read beyond the array is possible. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d1ccb6eaf9f..daf6975b0c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4684,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; - spec->slave_dig_outs[i - 1] = dig_nid; - if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) break; + spec->slave_dig_outs[i - 1] = dig_nid; } } if (spec->autocfg.dig_in_pin) @@ -9813,9 +9813,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; - spec->slave_dig_outs[i - 1] = dig_nid; - if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) break; + spec->slave_dig_outs[i - 1] = dig_nid; } } if (spec->autocfg.dig_in_pin) -- cgit v1.2.3-70-g09d2 From 46ef6ec9da420b298b1f197e445bf5b06fe01ef4 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 11 Nov 2009 14:32:10 -0500 Subject: ALSA: hda: Use model=mb5 for MacBookPro 5,2 BugLink: https://bugs.launchpad.net/bugs/462098 Until we can look closer at the verbs, let's use ALC885_MB5 for codec SSID 0x106b4600 to enable playback and capture for MacBookPro 5,2s. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index daf6975b0c2..84a52efdb2d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8911,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), - /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently - * no perfect solution yet + /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, + * so apparently no perfect solution yet */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- cgit v1.2.3-70-g09d2 From e2e527ae7fb07caa58f8fa8fa7e90ada0b175dd7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Nov 2009 08:28:03 +0100 Subject: ALSA: hda - Add another Nvidia HDMI codec id (10de:0005) Found on Nvidia 9800M GTS. Reported-by: Chris Balcum Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 9fb60276f5c..6afdab09bab 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -397,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, @@ -406,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { MODULE_ALIAS("snd-hda-codec-id:10de0002"); MODULE_ALIAS("snd-hda-codec-id:10de0003"); +MODULE_ALIAS("snd-hda-codec-id:10de0005"); MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); -- cgit v1.2.3-70-g09d2 From 5e08fe570c2dbabb5015c37049eb9a451e55c890 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 14 Nov 2009 14:37:19 +0100 Subject: ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume() Remove invlid __devinit prefix from the suspend callback. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/prodigy_hifi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c75515f5be6..6a9fee3ee78 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice) } #ifdef CONFIG_PM -static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +static int prodigy_hd2_resume(struct snd_ice1712 *ice) { /* initialize ak4396 codec and restore previous mixer volumes */ struct prodigy_hifi_spec *spec = ice->spec; -- cgit v1.2.3-70-g09d2 From 8ef5837a47f73faee18fa7ce2f9a9eb7675be8de Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sat, 14 Nov 2009 18:20:04 +0000 Subject: ALSA: hda - Dell Studio 1557 hd-audio quirk Add the Dell Studio 15 (model 1557, Core i7) laptop to the hd-audio quirk list, enabling audio. Signed-off-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8eb6508cd99..86de305fc9f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "Dell Studio 17", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be, "Dell Studio 1555", STAC_DELL_M6_DMIC), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, + "Dell Studio 1557", STAC_DELL_M6_DMIC), {} /* terminator */ }; -- cgit v1.2.3-70-g09d2 From e8e63cbf9a339c972eeb5ccf8777c8067bdfd869 Mon Sep 17 00:00:00 2001 From: Josh Triplett Date: Fri, 16 Oct 2009 16:03:49 -0700 Subject: oss: Mark loadhex static in hex2hex.c Nothing outside of hex2hex.c references loadhex. Signed-off-by: Josh Triplett --- sound/oss/hex2hex.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/hex2hex.c b/sound/oss/hex2hex.c index 5460faae98c..041ef5c52bc 100644 --- a/sound/oss/hex2hex.c +++ b/sound/oss/hex2hex.c @@ -12,7 +12,7 @@ #define MAX_SIZE (256*1024) unsigned char buf[MAX_SIZE]; -int loadhex(FILE *inf, unsigned char *buf) +static int loadhex(FILE *inf, unsigned char *buf) { int l=0, c, i; -- cgit v1.2.3-70-g09d2 From bf97402052483c125a9ea7bf13df0dd9b4134078 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 16 Nov 2009 11:07:17 +0200 Subject: ALSA: ice1724 - make some bitfields unsigned This is a clean up and doesn't change the behavior. Bit fields should always be unsigned. Otherwise pm_suspend_enabled will be -1 when you want it to be 1. The other bad thing is that the sparse checker will complain 36 times if they aren't unsigned. The other bitfields in that struct are unsigned already. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 9da2dae64c5..d063149e704 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -382,8 +382,8 @@ struct snd_ice1712 { #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); - int pm_suspend_enabled:1; - int pm_saved_is_spdif_master:1; + unsigned int pm_suspend_enabled:1; + unsigned int pm_saved_is_spdif_master:1; unsigned int pm_saved_spdif_ctrl; unsigned char pm_saved_spdif_cfg; unsigned int pm_saved_route; -- cgit v1.2.3-70-g09d2 From 0c3cee57efcb1c79d62b1238c0d22afef4599247 Mon Sep 17 00:00:00 2001 From: Javier Kohen Date: Tue, 17 Nov 2009 15:36:13 +0100 Subject: ALSA: usb - Quirk to disable master volume control in PCM2702 Disable the master volume control in the PCM2702 chipset. The datasheet documents two independent channel volume controls, one master mute control and one master volume control. All controls are fully functional except for the master volume control, which returns USB stalls on all GET requests. Signed-off-by: Javier Kohen Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 9efcfd08d74..c998220b99c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig channels = (ftr[0] - 7) / csize - 1; master_bits = snd_usb_combine_bytes(ftr + 6, csize); + /* master configuration quirks */ + switch (state->chip->usb_id) { + case USB_ID(0x08bb, 0x2702): + snd_printk(KERN_INFO + "usbmixer: master volume quirk for PCM2702 chip\n"); + /* disable non-functional volume control */ + master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1)); + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); else -- cgit v1.2.3-70-g09d2 From 12929baea4b29d70525f764034b3dac771dd69e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Nov 2009 15:58:35 +0100 Subject: ALSA: hda - Fix quirk for VAIO type G Vaio type G laptop doesn't work with the current quirk setup. After some tests, it turned out that it should be model=auto as default. Reported-by: Mattia Dongili Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 84a52efdb2d..70583719282 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11462,6 +11462,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", -- cgit v1.2.3-70-g09d2 From bd6ddcb41d5fbdcbc1486f48d8023f234b4a7f8d Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Tue, 17 Nov 2009 21:43:42 +0530 Subject: ASoC: Modifying the license string GPLv2 for OMAP3 EVM Correcting the license string from GPLv2 -> GPL v2. Found the problem while building OMAP3 ASoC driver as module. Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c263077..13aa380de16 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal "); MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-70-g09d2 From f3dd70414cdc0203ca63eef83ca130c2d1903b30 Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Sat, 7 Nov 2009 23:16:12 +0200 Subject: ASoC: OMAP3 Pandora: update for TWL4030 codec changes A while ago TWL4030 had it's playback stream name changed, but pandora needs it for it's playback path. Update to correct stream name so that playback works again. Also mark VIBRA output as not connected. Signed-off-by: Grazvydas Ignotas Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aaf7cb..0cd06f5dd35 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); snd_soc_dapm_nc_pin(codec, "HFL"); snd_soc_dapm_nc_pin(codec, "HFR"); + snd_soc_dapm_nc_pin(codec, "VIBRA"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); -- cgit v1.2.3-70-g09d2 From bab0212467e58929470ae3ae32515f17e30c3926 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Tue, 17 Nov 2009 13:51:01 -0700 Subject: ASoC: tlv320aic23 fix rate selection Fix the ordering of sr_valid_mask array. The lower bit of the index represents USB not bosr. Reported-by: Anuj Aggarwal Signed-off-by: Troy Kisky Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5cd72..6b24d8bb02b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = { #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) static const unsigned short sr_valid_mask[] = { LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ - LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ LOWER_GROUP, /* Usb, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ UPPER_GROUP, /* Usb, bosr - 1*/ }; /* -- cgit v1.2.3-70-g09d2 From 50b6bce59d154b5db137907a5c0ed45a4e7a3829 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Nov 2009 13:11:53 +0000 Subject: ASoC: Fix suspend with active audio streams When we get a stream suspend event force the power down since otherwise the stream would remain marked as active. In future we'll probably want to make this stream-specific and add an interface to make the power down of other widgets optional in order to support leaving bypass paths active while suspending the processor. Cc: stable@kernel.org Reported-by: Joonyoung Shim Tested-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d89f6dc0090..66d4c165f99 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (!w->power_check) continue; - power = w->power_check(w); - if (power) - sys_power = 1; + /* If we're suspending then pull down all the + * power. */ + switch (event) { + case SND_SOC_DAPM_STREAM_SUSPEND: + power = 0; + break; + + default: + power = w->power_check(w); + if (power) + sys_power = 1; + break; + } if (w->power == power) continue; @@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_SUSPEND: + sys_power = 0; + break; case SND_SOC_DAPM_STREAM_NOP: sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + break; default: break; } -- cgit v1.2.3-70-g09d2 From e9ff5eb2ae018fe2298c68746c873bf828c6b10e Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Fri, 27 Nov 2009 17:40:58 +0530 Subject: ASoC: AIC23: Fixing infinite loop in resume path This patch fixes two issues: a) Infinite loop in resume function b) Writes to non-existing registers in resume function Cc: stable@kernel.org Signed-off-by: Anuj Aggarwal Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8bb02b..90a0264f753 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + for (reg = 0; reg < TLV320AIC23_RESET; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- cgit v1.2.3-70-g09d2 From 4acd57c3de62374fe5bb52e5cd24538190f4eab2 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 29 Nov 2009 16:39:52 +0000 Subject: ALSA: AACI: fix AC97 multiple-open bug Signed-off-by: Russell King Cc: Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f8213e2d..1cb7c282a1f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -504,6 +504,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, int err; aaci_pcm_hw_free(substream); + if (aacirun->pcm_open) { + snd_ac97_pcm_close(aacirun->pcm); + aacirun->pcm_open = 0; + } err = devdma_hw_alloc(NULL, substream, params_buffer_bytes(params)); -- cgit v1.2.3-70-g09d2 From 8ee763b9c82c6ca0a59a7271ce4fa29d7baf5c09 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 29 Nov 2009 16:39:59 +0000 Subject: ALSA: AACI: fix recording bug pcm->r[1].slots is the double rate slot information, not the capture information. For capture, 'pcm' will already be the capture ac97 pcm structure. Signed-off-by: Russell King Cc: Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1cb7c282a1f..6c160a038b2 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -521,7 +521,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, else err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), params_channels(params), - aacirun->pcm->r[1].slots); + aacirun->pcm->r[0].slots); if (err) goto out; -- cgit v1.2.3-70-g09d2