From 42b16b3fbb5ee4555f5dee6220f3ccaa6e1ebe47 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 17 Jan 2011 00:09:38 +0100 Subject: =?UTF-8?q?Kill=20off=20warning:=20=E2=80=98inline=E2=80=99=20is?= =?UTF-8?q?=20not=20at=20beginning=20of=20declaration?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix a bunch of warning: ‘inline’ is not at beginning of declaration messages when building a 'make allyesconfig' kernel with -Wextra. These warnings are trivial to kill, yet rather annoying when building with -Wextra. The more we can cut down on pointless crap like this the better (IMHO). A previous patch to do this for a 'allnoconfig' build has already been merged. This just takes the cleanup a little further. Signed-off-by: Jesper Juhl Signed-off-by: Jiri Kosina --- sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_core.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index cf46bba..ecb8f4d 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -211,7 +211,7 @@ static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma); //static void vortex_adbdma_stopfifo(vortex_t *vortex, int adbdma); static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma); static void vortex_adbdma_resumefifo(vortex_t * vortex, int adbdma); -static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma); +static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma); static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma); #ifndef CHIP_AU8810 @@ -219,7 +219,7 @@ static void vortex_wtdma_startfifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_stopfifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_pausefifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_resumefifo(vortex_t * vortex, int wtdma); -static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma); +static inline int vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma); #endif /* global stuff. */ diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 23f49f3..d43252a 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1249,7 +1249,7 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { } } -static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) +static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; int temp; @@ -1498,7 +1498,7 @@ static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) POS_SHIFT) & POS_MASK); } #endif -static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) +static inline int vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) { stream_t *dma = &vortex->dma_wt[wtdma]; int temp; -- cgit v1.1 From b60fb519d7977e606621af85585c3677fc290ef8 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 25 Jan 2011 15:52:33 +0000 Subject: ALSA: AACI: fix multiple IRQ claiming Claiming the IRQ each time a playback or capture interface is opened is wasteful; the second copy of the registered handler is identical to the first and just wastes resources. Track the number of opens and only register the handler when necessary. Signed-off-by: Russell King --- sound/arm/aaci.c | 24 +++++++++++++++--------- sound/arm/aaci.h | 2 ++ 2 files changed, 17 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 24d3013..65685af 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -370,7 +370,7 @@ static int __aaci_pcm_open(struct aaci *aaci, struct aaci_runtime *aacirun) { struct snd_pcm_runtime *runtime = substream->runtime; - int ret; + int ret = 0; aacirun->substream = substream; runtime->private_data = aacirun; @@ -391,14 +391,15 @@ static int __aaci_pcm_open(struct aaci *aaci, */ runtime->hw.fifo_size = aaci->fifosize * 2; - ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED, - DRIVER_NAME, aaci); - if (ret) - goto out; - - return 0; + mutex_lock(&aaci->irq_lock); + if (!aaci->users++) { + ret = request_irq(aaci->dev->irq[0], aaci_irq, + IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci); + if (ret != 0) + aaci->users--; + } + mutex_unlock(&aaci->irq_lock); - out: return ret; } @@ -414,7 +415,11 @@ static int aaci_pcm_close(struct snd_pcm_substream *substream) WARN_ON(aacirun->cr & CR_EN); aacirun->substream = NULL; - free_irq(aaci->dev->irq[0], aaci); + + mutex_lock(&aaci->irq_lock); + if (!--aaci->users) + free_irq(aaci->dev->irq[0], aaci); + mutex_unlock(&aaci->irq_lock); return 0; } @@ -943,6 +948,7 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) aaci = card->private_data; mutex_init(&aaci->ac97_sem); + mutex_init(&aaci->irq_lock); aaci->card = card; aaci->dev = dev; diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 6a4a2ee..04c4568 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -226,6 +226,8 @@ struct aaci { struct snd_card *card; void __iomem *base; unsigned int fifosize; + unsigned int users; + struct mutex irq_lock; /* AC'97 */ struct mutex ac97_sem; -- cgit v1.1 From e831d80b453a3586f1e1664a705c153a4ced39b8 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 13 Jan 2011 10:13:17 +0000 Subject: ALSA: AACI: fix number of channels for record AC'97 codecs only support two channels for recording, so we shouldn't advertize that there are up to six channels available. Limit the selection of 4 and 6 channel audio to playback only. As this adds additional SNDRV_PCM_STREAM_PLAYBACK conditionals, we can combine some resulting in the elimination of __aaci_pcm_open() entirely, and making the code easier to read. Signed-off-by: Russell King --- sound/arm/aaci.c | 114 +++++++++++++++++++++++++------------------------------ 1 file changed, 52 insertions(+), 62 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 65685af..ab66d46 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -357,7 +357,7 @@ static struct snd_pcm_hardware aaci_hw_info = { /* rates are setup from the AC'97 codec */ .channels_min = 2, - .channels_max = 6, + .channels_max = 2, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 256, .period_bytes_max = PAGE_SIZE, @@ -365,22 +365,67 @@ static struct snd_pcm_hardware aaci_hw_info = { .periods_max = PAGE_SIZE / 16, }; -static int __aaci_pcm_open(struct aaci *aaci, - struct snd_pcm_substream *substream, - struct aaci_runtime *aacirun) +/* + * We can support two and four channel audio. Unfortunately + * six channel audio requires a non-standard channel ordering: + * 2 -> FL(3), FR(4) + * 4 -> FL(3), FR(4), SL(7), SR(8) + * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required) + * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual) + * This requires an ALSA configuration file to correct. + */ +static int aaci_rule_channels(struct snd_pcm_hw_params *p, + struct snd_pcm_hw_rule *rule) +{ + static unsigned int channel_list[] = { 2, 4, 6 }; + struct aaci *aaci = rule->private; + unsigned int mask = 1 << 0, slots; + + /* pcms[0] is the our 5.1 PCM instance. */ + slots = aaci->ac97_bus->pcms[0].r[0].slots; + if (slots & (1 << AC97_SLOT_PCM_SLEFT)) { + mask |= 1 << 1; + if (slots & (1 << AC97_SLOT_LFE)) + mask |= 1 << 2; + } + + return snd_interval_list(hw_param_interval(p, rule->var), + ARRAY_SIZE(channel_list), channel_list, mask); +} + +static int aaci_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct aaci *aaci = substream->private_data; + struct aaci_runtime *aacirun; int ret = 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aacirun = &aaci->playback; + } else { + aacirun = &aaci->capture; + } + aacirun->substream = substream; runtime->private_data = aacirun; runtime->hw = aaci_hw_info; runtime->hw.rates = aacirun->pcm->rates; snd_pcm_limit_hw_rates(runtime); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - aacirun->pcm->r[1].slots) - snd_ac97_pcm_double_rate_rules(runtime); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.channels_max = 6; + + /* Add rule describing channel dependency. */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + aaci_rule_channels, aaci, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + + if (aacirun->pcm->r[1].slots) + snd_ac97_pcm_double_rate_rules(runtime); + } /* * FIXME: ALSA specifies fifo_size in bytes. If we're in normal @@ -512,61 +557,6 @@ static const u32 channels_to_txmask[] = { [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9, }; -/* - * We can support two and four channel audio. Unfortunately - * six channel audio requires a non-standard channel ordering: - * 2 -> FL(3), FR(4) - * 4 -> FL(3), FR(4), SL(7), SR(8) - * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required) - * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual) - * This requires an ALSA configuration file to correct. - */ -static unsigned int channel_list[] = { 2, 4, 6 }; - -static int -aaci_rule_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule) -{ - struct aaci *aaci = rule->private; - unsigned int chan_mask = 1 << 0, slots; - - /* - * pcms[0] is the our 5.1 PCM instance. - */ - slots = aaci->ac97_bus->pcms[0].r[0].slots; - if (slots & (1 << AC97_SLOT_PCM_SLEFT)) { - chan_mask |= 1 << 1; - if (slots & (1 << AC97_SLOT_LFE)) - chan_mask |= 1 << 2; - } - - return snd_interval_list(hw_param_interval(p, rule->var), - ARRAY_SIZE(channel_list), channel_list, - chan_mask); -} - -static int aaci_pcm_open(struct snd_pcm_substream *substream) -{ - struct aaci *aaci = substream->private_data; - int ret; - - /* - * Add rule describing channel dependency. - */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - aaci_rule_channels, aaci, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (ret) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = __aaci_pcm_open(aaci, substream, &aaci->playback); - } else { - ret = __aaci_pcm_open(aaci, substream, &aaci->capture); - } - return ret; -} - static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { -- cgit v1.1 From 58e8a4741b519910cdabdd55c23f258e40cf6a3a Mon Sep 17 00:00:00 2001 From: Russell King Date: Wed, 26 Jan 2011 16:59:39 +0000 Subject: ALSA: AACI: fix channel mask selection When double-rate mode was selected, we weren't setting the additional two channel mask bits to allow double-rate to work. Rearrange the hw_params code to allow the correct channel mask to be selected. Signed-off-by: Russell King --- sound/arm/aaci.c | 70 ++++++++++++++++---------------------------------------- 1 file changed, 20 insertions(+), 50 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index ab66d46..8915e56 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -490,12 +490,21 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } +/* Channel to slot mask */ +static const u32 channels_to_slotmask[] = { + [2] = CR_SL3 | CR_SL4, + [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8, + [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9, +}; + static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, - struct aaci_runtime *aacirun, struct snd_pcm_hw_params *params) { + struct aaci_runtime *aacirun = substream->runtime->private_data; + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + int dbl = rate > 48000; int err; - struct aaci *aaci = substream->private_data; aaci_pcm_hw_free(substream); if (aacirun->pcm_open) { @@ -503,18 +512,21 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aacirun->pcm_open = 0; } + /* channels is already limited to 2, 4, or 6 by aaci_rule_channels */ + if (dbl && channels != 2) + return -EINVAL; + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (err >= 0) { - unsigned int rate = params_rate(params); - int dbl = rate > 48000; + struct aaci *aaci = substream->private_data; - err = snd_ac97_pcm_open(aacirun->pcm, rate, - params_channels(params), + err = snd_ac97_pcm_open(aacirun->pcm, rate, channels, aacirun->pcm->r[dbl].slots); aacirun->pcm_open = err == 0; aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; + aacirun->cr |= channels_to_slotmask[channels + dbl * 2]; aacirun->fifosz = aaci->fifosize * 4; if (aacirun->cr & CR_COMPACT) @@ -551,34 +563,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream) /* * Playback specific ALSA stuff */ -static const u32 channels_to_txmask[] = { - [2] = CR_SL3 | CR_SL4, - [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8, - [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9, -}; - -static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct aaci_runtime *aacirun = substream->runtime->private_data; - unsigned int channels = params_channels(params); - int ret; - - WARN_ON(channels >= ARRAY_SIZE(channels_to_txmask) || - !channels_to_txmask[channels]); - - ret = aaci_pcm_hw_params(substream, aacirun, params); - - /* - * Enable FIFO, compact mode, 16 bits per sample. - * FIXME: double rate slots? - */ - if (ret >= 0) - aacirun->cr |= channels_to_txmask[channels]; - - return ret; -} - static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun) { u32 ie; @@ -648,27 +632,13 @@ static struct snd_pcm_ops aaci_playback_ops = { .open = aaci_pcm_open, .close = aaci_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = aaci_pcm_playback_hw_params, + .hw_params = aaci_pcm_hw_params, .hw_free = aaci_pcm_hw_free, .prepare = aaci_pcm_prepare, .trigger = aaci_pcm_playback_trigger, .pointer = aaci_pcm_pointer, }; -static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct aaci_runtime *aacirun = substream->runtime->private_data; - int ret; - - ret = aaci_pcm_hw_params(substream, aacirun, params); - if (ret >= 0) - /* Line in record: slot 3 and 4 */ - aacirun->cr |= CR_SL3 | CR_SL4; - - return ret; -} - static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun) { u32 ie; @@ -765,7 +735,7 @@ static struct snd_pcm_ops aaci_capture_ops = { .open = aaci_pcm_open, .close = aaci_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = aaci_pcm_capture_hw_params, + .hw_params = aaci_pcm_hw_params, .hw_free = aaci_pcm_hw_free, .prepare = aaci_pcm_capture_prepare, .trigger = aaci_pcm_capture_trigger, -- cgit v1.1 From f006d8fc53c461aa66a9265597494f83ddf4f53d Mon Sep 17 00:00:00 2001 From: Russell King Date: Wed, 12 Jan 2011 23:46:03 +0000 Subject: ALSA: AACI: clean up AACI announcement printk Make the AACI announcement printk say which primecell part number has been found. Display the revision as an unsigned decimal, and display only the first 8 hex digits of the base address unless it's larger. Signed-off-by: Russell King --- sound/arm/aaci.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 8915e56..1e6d5f6 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -902,9 +902,9 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), - "%s at 0x%016llx, irq %d", - card->shortname, (unsigned long long)dev->res.start, - dev->irq[0]); + "%s PL%03x rev%u at 0x%08llx, irq %d", + card->shortname, amba_part(dev), amba_rev(dev), + (unsigned long long)dev->res.start, dev->irq[0]); aaci = card->private_data; mutex_init(&aaci->ac97_sem); -- cgit v1.1 From c0dea82c3c141c33ca22ca85f80e592028840864 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 13 Jan 2011 00:34:08 +0000 Subject: ALSA: AACI: use snd_pcm_lib_period_bytes() Use the helper rather than open-coding this. Signed-off-by: Russell King --- sound/arm/aaci.c | 4 ++-- sound/arm/aaci.h | 3 ++- 2 files changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1e6d5f6..a8f9538 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -541,11 +541,11 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct aaci_runtime *aacirun = runtime->private_data; + aacirun->period = snd_pcm_lib_period_bytes(substream); aacirun->start = runtime->dma_area; aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; - aacirun->period = - aacirun->bytes = frames_to_bytes(runtime, runtime->period_size); + aacirun->bytes = aacirun->period; return 0; } diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 04c4568..198750d 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -210,6 +210,8 @@ struct aaci_runtime { u32 cr; struct snd_pcm_substream *substream; + unsigned int period; /* byte size of a "period" */ + /* * PIO support */ @@ -217,7 +219,6 @@ struct aaci_runtime { void *end; void *ptr; int bytes; - unsigned int period; unsigned int fifosz; }; -- cgit v1.1 From ea51d0b164040ad594c1f9c4c6faf23c19c977b9 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 13 Jan 2011 08:47:35 +0000 Subject: ALSA: AACI: no need to call snd_pcm_period_elapsed() for each period There is no need to call snd_pcm_period_elapsed() each time a period elapses - we can call it after we're done once loading/unloading the FIFO with data. ALSA works out how many periods have elapsed by reading the current pointers. Signed-off-by: Russell King --- sound/arm/aaci.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index a8f9538..393ce08 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -206,6 +206,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (mask & ISR_RXINTR) { struct aaci_runtime *aacirun = &aaci->capture; + bool period_elapsed = false; void *ptr; if (!aacirun->substream || !aacirun->start) { @@ -223,10 +224,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (aacirun->bytes <= 0) { aacirun->bytes += aacirun->period; - aacirun->ptr = ptr; - spin_unlock(&aacirun->lock); - snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aacirun->lock); + period_elapsed = true; } if (!(aacirun->cr & CR_EN)) break; @@ -256,6 +254,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) aacirun->ptr = ptr; spin_unlock(&aacirun->lock); + + if (period_elapsed) + snd_pcm_period_elapsed(aacirun->substream); } if (mask & ISR_URINTR) { @@ -265,6 +266,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (mask & ISR_TXINTR) { struct aaci_runtime *aacirun = &aaci->playback; + bool period_elapsed = false; void *ptr; if (!aacirun->substream || !aacirun->start) { @@ -282,10 +284,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (aacirun->bytes <= 0) { aacirun->bytes += aacirun->period; - aacirun->ptr = ptr; - spin_unlock(&aacirun->lock); - snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aacirun->lock); + period_elapsed = true; } if (!(aacirun->cr & CR_EN)) break; @@ -315,6 +314,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) aacirun->ptr = ptr; spin_unlock(&aacirun->lock); + + if (period_elapsed) + snd_pcm_period_elapsed(aacirun->substream); } } -- cgit v1.1 From 5d350cba486de34eff99d0394d8fb436af54522e Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 13 Jan 2011 22:25:10 +0000 Subject: ALSA: AACI: make fifo variables more explanitory Improve commenting and change fifo variable names to reflect their meanings. Signed-off-by: Russell King --- sound/arm/aaci.c | 42 ++++++++++++++++++++++++------------------ sound/arm/aaci.h | 4 ++-- 2 files changed, 26 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 393ce08..a148e27 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -219,7 +219,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) ptr = aacirun->ptr; do { - unsigned int len = aacirun->fifosz; + unsigned int len = aacirun->fifo_bytes; u32 val; if (aacirun->bytes <= 0) { @@ -279,7 +279,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) ptr = aacirun->ptr; do { - unsigned int len = aacirun->fifosz; + unsigned int len = aacirun->fifo_bytes; u32 val; if (aacirun->bytes <= 0) { @@ -430,13 +430,11 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) } /* - * FIXME: ALSA specifies fifo_size in bytes. If we're in normal - * mode, each 32-bit word contains one sample. If we're in - * compact mode, each 32-bit word contains two samples, effectively - * halving the FIFO size. However, we don't know for sure which - * we'll be using at this point. We set this to the lower limit. + * ALSA wants the byte-size of the FIFOs. As we only support + * 16-bit samples, this is twice the FIFO depth irrespective + * of whether it's in compact mode or not. */ - runtime->hw.fifo_size = aaci->fifosize * 2; + runtime->hw.fifo_size = aaci->fifo_depth * 2; mutex_lock(&aaci->irq_lock); if (!aaci->users++) { @@ -529,10 +527,13 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aacirun->pcm_open = err == 0; aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; aacirun->cr |= channels_to_slotmask[channels + dbl * 2]; - aacirun->fifosz = aaci->fifosize * 4; - if (aacirun->cr & CR_COMPACT) - aacirun->fifosz >>= 1; + /* + * fifo_bytes is the number of bytes we transfer to/from + * the FIFO, including padding. So that's x4. As we're + * in compact mode, the FIFO is half the size. + */ + aacirun->fifo_bytes = aaci->fifo_depth * 4 / 2; } return err; @@ -948,6 +949,10 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) struct aaci_runtime *aacirun = &aaci->playback; int i; + /* + * Enable the channel, but don't assign it to any slots, so + * it won't empty onto the AC'97 link. + */ writel(CR_FEN | CR_SZ16 | CR_EN, aacirun->base + AACI_TXCR); for (i = 0; !(readl(aacirun->base + AACI_SR) & SR_TXFF) && i < 4096; i++) @@ -964,7 +969,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) writel(aaci->maincr, aaci->base + AACI_MAINCR); /* - * If we hit 4096, we failed. Go back to the specified + * If we hit 4096 entries, we failed. Go back to the specified * fifo depth. */ if (i == 4096) @@ -1029,11 +1034,12 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) /* * Size the FIFOs (must be multiple of 16). + * This is the number of entries in the FIFO. */ - aaci->fifosize = aaci_size_fifo(aaci); - if (aaci->fifosize & 15) { - printk(KERN_WARNING "AACI: fifosize = %d not supported\n", - aaci->fifosize); + aaci->fifo_depth = aaci_size_fifo(aaci); + if (aaci->fifo_depth & 15) { + printk(KERN_WARNING "AACI: FIFO depth %d not supported\n", + aaci->fifo_depth); ret = -ENODEV; goto out; } @@ -1046,8 +1052,8 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) ret = snd_card_register(aaci->card); if (ret == 0) { - dev_info(&dev->dev, "%s, fifo %d\n", aaci->card->longname, - aaci->fifosize); + dev_info(&dev->dev, "%s\n", aaci->card->longname); + dev_info(&dev->dev, "FIFO %u entries\n", aaci->fifo_depth); amba_set_drvdata(dev, aaci->card); return ret; } diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 198750d..5791bd5 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -219,14 +219,14 @@ struct aaci_runtime { void *end; void *ptr; int bytes; - unsigned int fifosz; + unsigned int fifo_bytes; }; struct aaci { struct amba_device *dev; struct snd_card *card; void __iomem *base; - unsigned int fifosize; + unsigned int fifo_depth; unsigned int users; struct mutex irq_lock; -- cgit v1.1 From 59b479e0985f0b795d68331d6443a7f89c47768d Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Thu, 27 Jan 2011 16:39:40 -0800 Subject: omap: Start using CONFIG_SOC_OMAP We want to have just CONFIG_ARCH_OMAP2, 3 and 4. The rest are nowadays just subcategories of these. Search and replace the following: ARCH_OMAP2420 SOC_OMAP2420 ARCH_OMAP2430 SOC_OMAP2430 ARCH_OMAP3430 SOC_OMAP3430 No functional changes. Signed-off-by: Tony Lindgren Signed-off-by: Thomas Weber Acked-by: Sourav Poddar --- sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-mcbsp.h | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d203f4d..ede6afd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -92,7 +92,7 @@ static const unsigned long omap1_mcbsp_port[][2] = {}; static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) +#if defined(CONFIG_SOC_OMAP2430) || defined(CONFIG_ARCH_OMAP3) { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, @@ -113,7 +113,7 @@ static const int omap44xx_dma_reqs[][2] = { static const int omap44xx_dma_reqs[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2420) +#if defined(CONFIG_SOC_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, @@ -124,7 +124,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = { static const unsigned long omap2420_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2430) +#if defined(CONFIG_SOC_OMAP2430) static const unsigned long omap2430_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 110c106..37dc721 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -43,7 +43,7 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -#if defined(CONFIG_ARCH_OMAP2420) +#if defined(CONFIG_SOC_OMAP2420) #define NUM_LINKS 2 #endif #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) @@ -54,7 +54,7 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 4 #endif -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) +#if defined(CONFIG_ARCH_OMAP3) || defined(CONFIG_SOC_OMAP2430) #undef NUM_LINKS #define NUM_LINKS 5 #endif -- cgit v1.1 From b4a5660da011a0d55cac2ead05a9171d5544d272 Mon Sep 17 00:00:00 2001 From: Kukjin Kim Date: Mon, 14 Feb 2011 16:53:58 +0900 Subject: ASoC: Change dependency of ARCH_EXYNOS4 This patch changes dependency of ARCH_EXYNOS4 from ARCH_S5PV310 according to the change of ARCH name, EXYNOS4. Acked-by: Jassi Brar Cc: Liam Girdwood Acked-by: Mark Brown Signed-off-by: Kukjin Kim --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a6a6b5f..d6713d5 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_S5PV310 + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_EXYNOS4 select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C2410 help -- cgit v1.1 From aa25afad2ca60d19457849ea75e9c31236f4e174 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sat, 19 Feb 2011 15:55:00 +0000 Subject: ARM: amba: make probe() functions take const id tables Make Primecell driver probe functions take a const pointer to their ID tables. Drivers should never modify their ID tables in their probe handler. Signed-off-by: Russell King --- sound/arm/aaci.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7c1fc64..d0821f8 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1011,7 +1011,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) return i; } -static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) +static int __devinit aaci_probe(struct amba_device *dev, + const struct amba_id *id) { struct aaci *aaci; int ret, i; -- cgit v1.1 From 2686e07b3efe45d964acba22c3a756fd7dd81389 Mon Sep 17 00:00:00 2001 From: Kishon Vijay Abraham I Date: Thu, 24 Feb 2011 15:16:56 +0530 Subject: ASoC: McBSP: get hw params from McBSP driver Removed the use of macros to obtain base address and DMA channel number. Instead use the McBSP driver API's that passes base address and DMA channel number to the client driver. Signed-off-by: Kishon Vijay Abraham I Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Tony Lindgren --- sound/soc/omap/omap-mcbsp.c | 126 ++------------------------------------------ 1 file changed, 4 insertions(+), 122 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index ede6afd..2175f09 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -69,110 +69,6 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; */ static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; -#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) -static const int omap1_dma_reqs[][2] = { - { OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX }, - { OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX }, - { OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX }, -}; -static const unsigned long omap1_mcbsp_port[][2] = { - { OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, - OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, - { OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1, - OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, - { OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1, - OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 }, -}; -#else -static const int omap1_dma_reqs[][2] = {}; -static const unsigned long omap1_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3) -static const int omap24xx_dma_reqs[][2] = { - { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, - { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, -#if defined(CONFIG_SOC_OMAP2430) || defined(CONFIG_ARCH_OMAP3) - { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, - { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, - { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, -#endif -}; -#else -static const int omap24xx_dma_reqs[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP4) -static const int omap44xx_dma_reqs[][2] = { - { OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX }, - { OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX }, - { OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX }, - { OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX }, -}; -#else -static const int omap44xx_dma_reqs[][2] = {}; -#endif - -#if defined(CONFIG_SOC_OMAP2420) -static const unsigned long omap2420_mcbsp_port[][2] = { - { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, - OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, - { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1, - OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, -}; -#else -static const unsigned long omap2420_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_SOC_OMAP2430) -static const unsigned long omap2430_mcbsp_port[][2] = { - { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, - OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, - OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, - OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, - OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, - OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, -}; -#else -static const unsigned long omap2430_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP3) -static const unsigned long omap34xx_mcbsp_port[][2] = { - { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, -}; -#else -static const unsigned long omap34xx_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP4) -static const unsigned long omap44xx_mcbsp_port[][2] = { - { OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, -}; -#else -static const unsigned long omap44xx_mcbsp_port[][2] = {}; -#endif - static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -346,24 +242,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, unsigned int format, div, framesize, master; dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream]; - if (cpu_class_is_omap1()) { - dma = omap1_dma_reqs[bus_id][substream->stream]; - port = omap1_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap2420()) { - dma = omap24xx_dma_reqs[bus_id][substream->stream]; - port = omap2420_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap2430()) { - dma = omap24xx_dma_reqs[bus_id][substream->stream]; - port = omap2430_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap343x()) { - dma = omap24xx_dma_reqs[bus_id][substream->stream]; - port = omap34xx_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap44xx()) { - dma = omap44xx_dma_reqs[bus_id][substream->stream]; - port = omap44xx_mcbsp_port[bus_id][substream->stream]; - } else { - return -ENODEV; - } + + dma = omap_mcbsp_dma_ch_params(bus_id, substream->stream); + port = omap_mcbsp_dma_reg_params(bus_id, substream->stream); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: dma_data->data_type = OMAP_DMA_DATA_TYPE_S16; -- cgit v1.1 From f07eb223a081b278be02a58394cb5fd66f1a1bbd Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Tue, 22 Feb 2011 21:05:04 -0700 Subject: dt/sound: Eliminate users of of_platform_{,un}register_driver Get rid of users of of_platform_driver in drivers/sound. The of_platform_{,un}register_driver functions are going away, so the users need to be converted to using the platform_bus_type directly. Signed-off-by: Grant Likely --- sound/soc/fsl/fsl_dma.c | 9 ++++----- sound/soc/fsl/fsl_ssi.c | 9 ++++----- sound/soc/fsl/mpc5200_dma.c | 24 +++++++++++------------- sound/soc/fsl/mpc5200_psc_ac97.c | 9 ++++----- sound/soc/fsl/mpc5200_psc_i2s.c | 9 ++++----- sound/sparc/amd7930.c | 8 ++++---- sound/sparc/cs4231.c | 16 ++++++++-------- sound/sparc/dbri.c | 8 ++++---- 8 files changed, 43 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 4cf98c0..15dac0f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -896,8 +896,7 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -static int __devinit fsl_soc_dma_probe(struct platform_device *pdev, - const struct of_device_id *match) +static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) { struct dma_object *dma; struct device_node *np = pdev->dev.of_node; @@ -979,7 +978,7 @@ static const struct of_device_id fsl_soc_dma_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids); -static struct of_platform_driver fsl_soc_dma_driver = { +static struct platform_driver fsl_soc_dma_driver = { .driver = { .name = "fsl-pcm-audio", .owner = THIS_MODULE, @@ -993,12 +992,12 @@ static int __init fsl_soc_dma_init(void) { pr_info("Freescale Elo DMA ASoC PCM Driver\n"); - return of_register_platform_driver(&fsl_soc_dma_driver); + return platform_driver_register(&fsl_soc_dma_driver); } static void __exit fsl_soc_dma_exit(void) { - of_unregister_platform_driver(&fsl_soc_dma_driver); + platform_driver_unregister(&fsl_soc_dma_driver); } module_init(fsl_soc_dma_init); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4cc167a..313e0cc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -624,8 +624,7 @@ static void make_lowercase(char *s) } } -static int __devinit fsl_ssi_probe(struct platform_device *pdev, - const struct of_device_id *match) +static int __devinit fsl_ssi_probe(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private; int ret = 0; @@ -774,7 +773,7 @@ static const struct of_device_id fsl_ssi_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_ssi_ids); -static struct of_platform_driver fsl_ssi_driver = { +static struct platform_driver fsl_ssi_driver = { .driver = { .name = "fsl-ssi-dai", .owner = THIS_MODULE, @@ -788,12 +787,12 @@ static int __init fsl_ssi_init(void) { printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); - return of_register_platform_driver(&fsl_ssi_driver); + return platform_driver_register(&fsl_ssi_driver); } static void __exit fsl_ssi_exit(void) { - of_unregister_platform_driver(&fsl_ssi_driver); + platform_driver_unregister(&fsl_ssi_driver); } module_init(fsl_ssi_init); diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index f92dca0..fff695c 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -368,8 +368,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op, - const struct of_device_id *match) +static int mpc5200_hpcd_probe(struct of_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -511,32 +510,31 @@ static int mpc5200_hpcd_remove(struct of_device *op) } static struct of_device_id mpc5200_hpcd_match[] = { - { - .compatible = "fsl,mpc5200-pcm", - }, + { .compatible = "fsl,mpc5200-pcm", }, {} }; MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); -static struct of_platform_driver mpc5200_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc5200-pcm-audio", - .match_table = mpc5200_hpcd_match, +static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, + .dev = { + .owner = THIS_MODULE, + .name = "mpc5200-pcm-audio", + .of_match_table = mpc5200_hpcd_match, + } }; static int __init mpc5200_hpcd_init(void) { - return of_register_platform_driver(&mpc5200_hpcd_of_driver); + return platform_driver_register(&mpc5200_hpcd_of_driver); } +module_init(mpc5200_hpcd_init); static void __exit mpc5200_hpcd_exit(void) { - of_unregister_platform_driver(&mpc5200_hpcd_of_driver); + platform_driver_unregister(&mpc5200_hpcd_of_driver); } - -module_init(mpc5200_hpcd_init); module_exit(mpc5200_hpcd_exit); MODULE_AUTHOR("Grant Likely "); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 40acc8e..ad36b09 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -272,8 +272,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { * - Probe/remove operations * - OF device match table */ -static int __devinit psc_ac97_of_probe(struct platform_device *op, - const struct of_device_id *match) +static int __devinit psc_ac97_of_probe(struct platform_device *op) { int rc; struct snd_ac97 ac97; @@ -316,7 +315,7 @@ static struct of_device_id psc_ac97_match[] __devinitdata = { }; MODULE_DEVICE_TABLE(of, psc_ac97_match); -static struct of_platform_driver psc_ac97_driver = { +static struct platform_driver psc_ac97_driver = { .probe = psc_ac97_of_probe, .remove = __devexit_p(psc_ac97_of_remove), .driver = { @@ -332,13 +331,13 @@ static struct of_platform_driver psc_ac97_driver = { */ static int __init psc_ac97_init(void) { - return of_register_platform_driver(&psc_ac97_driver); + return platform_driver_register(&psc_ac97_driver); } module_init(psc_ac97_init); static void __exit psc_ac97_exit(void) { - of_unregister_platform_driver(&psc_ac97_driver); + platform_driver_unregister(&psc_ac97_driver); } module_exit(psc_ac97_exit); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9018fa5..87cf2a5 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -150,8 +150,7 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{ * - Probe/remove operations * - OF device match table */ -static int __devinit psc_i2s_of_probe(struct platform_device *op, - const struct of_device_id *match) +static int __devinit psc_i2s_of_probe(struct platform_device *op) { int rc; struct psc_dma *psc_dma; @@ -213,7 +212,7 @@ static struct of_device_id psc_i2s_match[] __devinitdata = { }; MODULE_DEVICE_TABLE(of, psc_i2s_match); -static struct of_platform_driver psc_i2s_driver = { +static struct platform_driver psc_i2s_driver = { .probe = psc_i2s_of_probe, .remove = __devexit_p(psc_i2s_of_remove), .driver = { @@ -229,13 +228,13 @@ static struct of_platform_driver psc_i2s_driver = { */ static int __init psc_i2s_init(void) { - return of_register_platform_driver(&psc_i2s_driver); + return platform_driver_register(&psc_i2s_driver); } module_init(psc_i2s_init); static void __exit psc_i2s_exit(void) { - of_unregister_platform_driver(&psc_i2s_driver); + platform_driver_unregister(&psc_i2s_driver); } module_exit(psc_i2s_exit); diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index f8bcfc3..ad7d4d7 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1002,7 +1002,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_sbus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit amd7930_sbus_probe(struct platform_device *op) { struct resource *rp = &op->resource[0]; static int dev_num; @@ -1064,7 +1064,7 @@ static const struct of_device_id amd7930_match[] = { {}, }; -static struct of_platform_driver amd7930_sbus_driver = { +static struct platform_driver amd7930_sbus_driver = { .driver = { .name = "audio", .owner = THIS_MODULE, @@ -1075,7 +1075,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - return of_register_platform_driver(&amd7930_sbus_driver); + return platform_driver_register(&amd7930_sbus_driver); } static void __exit amd7930_exit(void) @@ -1092,7 +1092,7 @@ static void __exit amd7930_exit(void) amd7930_list = NULL; - of_unregister_platform_driver(&amd7930_sbus_driver); + platform_driver_unregister(&amd7930_sbus_driver); } module_init(amd7930_init); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index c276086..0e618f8 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1856,7 +1856,7 @@ static int __devinit snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_sbus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_sbus_probe(struct platform_device *op) { struct resource *rp = &op->resource[0]; struct snd_card *card; @@ -2048,7 +2048,7 @@ static int __devinit snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_ebus_probe(struct platform_device *op) { struct snd_card *card; int err; @@ -2072,16 +2072,16 @@ static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct } #endif -static int __devinit cs4231_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_probe(struct platform_device *op) { #ifdef EBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "ebus")) - return cs4231_ebus_probe(op, match); + return cs4231_ebus_probe(op); #endif #ifdef SBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "sbus") || !strcmp(op->dev.of_node->parent->name, "sbi")) - return cs4231_sbus_probe(op, match); + return cs4231_sbus_probe(op); #endif return -ENODEV; } @@ -2108,7 +2108,7 @@ static const struct of_device_id cs4231_match[] = { MODULE_DEVICE_TABLE(of, cs4231_match); -static struct of_platform_driver cs4231_driver = { +static struct platform_driver cs4231_driver = { .driver = { .name = "audio", .owner = THIS_MODULE, @@ -2120,12 +2120,12 @@ static struct of_platform_driver cs4231_driver = { static int __init cs4231_init(void) { - return of_register_platform_driver(&cs4231_driver); + return platform_driver_register(&cs4231_driver); } static void __exit cs4231_exit(void) { - of_unregister_platform_driver(&cs4231_driver); + platform_driver_unregister(&cs4231_driver); } module_init(cs4231_init); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 39cd5d6..73f9cba 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2592,7 +2592,7 @@ static void snd_dbri_free(struct snd_dbri *dbri) (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit dbri_probe(struct platform_device *op) { struct snd_dbri *dbri; struct resource *rp; @@ -2686,7 +2686,7 @@ static const struct of_device_id dbri_match[] = { MODULE_DEVICE_TABLE(of, dbri_match); -static struct of_platform_driver dbri_sbus_driver = { +static struct platform_driver dbri_sbus_driver = { .driver = { .name = "dbri", .owner = THIS_MODULE, @@ -2699,12 +2699,12 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_platform_driver(&dbri_sbus_driver); + return platform_driver_register(&dbri_sbus_driver); } static void __exit dbri_exit(void) { - of_unregister_platform_driver(&dbri_sbus_driver); + platform_driver_unregister(&dbri_sbus_driver); } module_init(dbri_init); -- cgit v1.1 From 2c4066cca46e2e82b63127a3dba5e5f68a612749 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Eric=20B=C3=A9nard?= Date: Fri, 25 Feb 2011 13:48:15 +0100 Subject: eukrea-tlv320: add MBIMXSD51 support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Eric Bénard Signed-off-by: Sascha Hauer --- sound/soc/imx/Kconfig | 3 ++- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 642270a..9eeb8f0 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -44,7 +44,8 @@ config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD + || MACH_EUKREA_MBIMXSD35_BASEBOARD \ + || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index e20c9e1..a08e822 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -98,7 +98,8 @@ static int __init eukrea_tlv320_init(void) int ret; if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() - && !machine_is_eukrea_cpuimx35sd()) + && !machine_is_eukrea_cpuimx35sd() + && !machine_is_eukrea_cpuimx51sd()) /* return happy. We might run on a totally different machine */ return 0; -- cgit v1.1 From ffd6eae2a0d18ca4a741615292a9c9ce904307fb Mon Sep 17 00:00:00 2001 From: Abhilash K V Date: Tue, 8 Mar 2011 21:02:43 +0530 Subject: ASoC: AM3517: Update codec name after multi-component update The i2c client device name (".2-001a" in this case, including the separator period) for the AIC23 codec on the TI AM3517-EVM was appended to the codec_name member of am3517evm_dai to resolve the names mismatch happening in soc_bind_dai_link(), due to which the card was not getting registered. Signed-off-by: Abhilash K V Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/am3517evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 1617504..73dde4a 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = { .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", - .codec_name = "tlv320aic23-codec", + .codec_name = "tlv320aic23-codec.2-001a", .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; -- cgit v1.1 From a110f4ef810ee29d810876df725f41d66629733e Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Wed, 9 Mar 2011 21:46:20 +0100 Subject: ASoC: mini2440: Fix uda134x codec problem. ASoC audio for mini2440 platform in current kenrel doesn't work. First problem is samsung_asoc_dma device is missing in initialization. Next problem is with codec. Codec is initialized but never probed because no platform_device exist for codec driver. It leads to errors during codec binding to asoc dai. Next problem was platform data which was passed from board to asoc main driver but not passed to codec when called codec_soc_probe(). Following patch should fix issues. But not sure if in correct way. Please review. Signed-off-by: Marek Belisko Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/uda134x.c | 3 ++- sound/soc/samsung/s3c24xx_uda134x.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e76847a..48ffd40 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev); + struct uda134x_platform_data *pd = codec->card->dev->platform_data; + int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 3cb7007..dc9d551 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -219,7 +219,7 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = { static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .name = "UDA134X", .stream_name = "UDA134X", - .codec_name = "uda134x-hifi", + .codec_name = "uda134x-codec", .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, @@ -314,6 +314,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev) platform_set_drvdata(s3c24xx_uda134x_snd_device, &snd_soc_s3c24xx_uda134x); + platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x)); ret = platform_device_add(s3c24xx_uda134x_snd_device); if (ret) { printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); -- cgit v1.1 From b1a56b331aec59be04f25ac99694d855d591c539 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Mar 2011 18:18:53 +0000 Subject: ASoC: Remove bogus check for register validity in debugfs write Since not all registers need to be cached and the cache is entirely optional anyway we shouldn't be checking that a register is in the cached range. If the register is invalid then the actual I/O code can determine that and report an error. Similarly, the step size can and should be enforced by the lower level code if it's important. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 17efacd..4dda589 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->driver->reg_cache_size) || (reg % step)) - return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) -- cgit v1.1 From 40285f832b09feb621d8da9db7983200a4b29311 Mon Sep 17 00:00:00 2001 From: Matti Aaltonen Date: Tue, 1 Mar 2011 10:10:37 -0300 Subject: [media] ASoC: WL1273 FM radio: Access I2C IO functions through pointers These changes are needed to keep up with the changes in the MFD core and V4L2 parts of the wl1273 FM radio driver. Use function pointers instead of exported functions for I2C IO. Also move all preprocessor constants from the wl1273.h to include/linux/mfd/wl1273-core.h. Also update the year in the copyright statement. Signed-off-by: Matti J. Aaltonen Acked-by: Mark Brown Signed-off-by: Mauro Carvalho Chehab --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 11 ++++++++--- 2 files changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c48b23c..9726d6e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C - select SND_SOC_WL1273 if RADIO_WL1273 + select SND_SOC_WL1273 if MFD_WL1273_CORE select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f..5836201 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -3,7 +3,7 @@ * * Author: Matti Aaltonen, * - * Copyright: (C) 2010 Nokia Corporation + * Copyright: (C) 2010, 2011 Nokia Corporation * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -179,7 +179,12 @@ static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, return 0; } -static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; +/* + * TODO: Implement the audio routing in the driver. Now this control + * only indicates the setting that has been done elsewhere (in the user + * space). + */ +static const char * const wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -239,7 +244,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, return 1; } -static const char *wl1273_audio_strings[] = { "Digital", "Analog" }; +static const char * const wl1273_audio_strings[] = { "Digital", "Analog" }; static const struct soc_enum wl1273_audio_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), -- cgit v1.1 From bff5fbf50bd498c217994bd2d41a53ac3141185a Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:21:38 +0800 Subject: ALSA: hda - VIA: Fix stereo mixer recording no sound issue Modify function via_mux_enum_put() to fix stereo mixer recording no sound issue. Signed-off-by: Lydia Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 63b0054..5bc9bd9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1101,6 +1101,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int ret; if (!spec->mux_nids[adc_idx]) return -EINVAL; @@ -1109,12 +1110,14 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - /* update jack power state */ - set_jack_power_state(codec); - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); + /* update jack power state */ + set_jack_power_state(codec); + + return ret; } static int via_independent_hp_info(struct snd_kcontrol *kcontrol, -- cgit v1.1 From ce0e5a9e81fbb153ee15ca60246c6722f07fc546 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:22:37 +0800 Subject: ALSA: hda - VIA: Fix independent headphone no sound issue Modify via_independent_hp_put() function to support VT1718S and VT1812 codecs, and fix independent headphone no sound issue. Signed-off-by: Lydia Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 5bc9bd9..2f605e5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1191,8 +1191,16 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1718S) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + else + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -1211,6 +1219,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, activate_ctl(codec, "Headphone Playback Switch", spec->hp_independent_mode); } + /* update jack power state */ + set_jack_power_state(codec); return 0; } -- cgit v1.1 From ab657e0cacc39d88145871c6a3c844597c02d406 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:23:23 +0800 Subject: ALSA: hda - VIA: Add missing support for VT1718S in A-A path Modify mute_aa_path() function to support VT1718S codec. Signed-off-by: Lydia Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2f605e5..23e58f0 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1323,6 +1323,11 @@ static void mute_aa_path(struct hda_codec *codec, int mute) start_idx = 2; end_idx = 4; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return; } -- cgit v1.1 From 169222813eec8403c76394fb7b35ecab98e3c607 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:24:10 +0800 Subject: ALSA: hda - VIA: Fix invalid A-A path volume adjust issue Modify vt_auto_create_analog_input_ctls() function to fix invalid a-a path volume adjust issue for VT1708S, VT1702 and VT1716S codecs. Signed-off-by: Lydia Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 23e58f0..299a18b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2456,7 +2456,14 @@ static int vt_auto_create_analog_input_ctls(struct hda_codec *codec, else type_idx = 0; label = hda_get_autocfg_input_label(codec, cfg, i); - err = via_new_analog_input(spec, label, type_idx, idx, cap_nid); + if (spec->codec_type == VT1708S || + spec->codec_type == VT1702 || + spec->codec_type == VT1716S) + err = via_new_analog_input(spec, label, type_idx, + idx+1, cap_nid); + else + err = via_new_analog_input(spec, label, type_idx, + idx, cap_nid); if (err < 0) return err; snd_hda_add_imux_item(imux, label, idx, NULL); -- cgit v1.1 From 0341ccd7557fecafe6a79c55158670cf629d269e Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:25:03 +0800 Subject: ALSA: hda - VIA: Fix codec type for VT1708BCE at the right timing Add get_codec_type() in via_new_spec() function to make sure getting correct codec type before building mixer controls. Signed-off-by: Lydia Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 299a18b..269bb36 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -159,6 +159,7 @@ struct via_spec { #endif }; +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); static struct via_spec * via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -169,6 +170,10 @@ static struct via_spec * via_new_spec(struct hda_codec *codec) codec->spec = spec; spec->codec = codec; + spec->codec_type = get_codec_type(codec); + /* VT1708BCE & VT1708S are almost same */ + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; return spec; } @@ -2203,10 +2208,6 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); - spec->codec_type = get_codec_type(codec); - if (spec->codec_type == VT1708BCE) - spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost - same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { -- cgit v1.1 From 970f630f5adcefb2841338929e209d970001d919 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:25:56 +0800 Subject: ALSA: hda - VIA: Correct stream names for VT1818S Correct stream names of analog playback and capture streams for VT1818S. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 269bb36..7e317f9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4173,6 +4173,11 @@ static int patch_vt1708S(struct hda_codec *codec) spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + /* correct names for VT1818S */ + if (codec->vendor_id == 0x11060440) { + spec->stream_name_analog = "VT1818S Analog"; + spec->stream_name_digital = "VT1818S Digital"; + } return 0; } -- cgit v1.1 From ee3c35c0827de02de414d08b2ddcbb910c2263ab Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Tue, 22 Mar 2011 16:26:36 +0800 Subject: ALSA: hda - VIA: Fix VT1708 can't build up Headphone control issue Since VT1708 didn't support the control of getting connection number, building of headphone control will fail in via_hp_build() function. Signed-off-by: Lydia Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 7e317f9..1371b57 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1266,9 +1266,12 @@ static int via_hp_build(struct hda_codec *codec) break; } - nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); - if (nums <= 1) - return 0; + if (spec->codec_type != VT1708) { + nums = snd_hda_get_connections(codec, nid, + conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + } knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL) -- cgit v1.1 From 333802e90d3f0366c4a1cb767e2783d2e1df73a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Mar 2011 12:02:33 +0000 Subject: ASoC: Support !REGULATOR build for sgtl5000 The regulator is optional depending on board design. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/sgtl5000.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f7217f..ff29380 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_REGULATOR static int ldo_regulator_is_enabled(struct regulator_dev *dev) { struct ldo_regulator *ldo = rdev_get_drvdata(dev); @@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) return 0; } +#else +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + return -EINVAL; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + return 0; +} +#endif /* * set dac bias -- cgit v1.1 From 5a8826463c19b0d1a2fc60b2adac0ec318047844 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 23 Mar 2011 08:35:07 +0100 Subject: ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side Similar to commit 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4, this patch avoids unnecessary volume control indices for more Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side" controls. These indices cause these volume controls to be ignored by PulseAudio and vmaster and should be removed whenever possible. Cc: stable@kernel.org Reported-by: Jan Losinski Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28f95d1..5d582de 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16008,9 +16008,12 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc861_create_out_sw(codec, name, nid, i, 3); + index = 0; + } + err = __alc861_create_out_sw(codec, name, nid, index, 3); if (err < 0) return err; } @@ -17161,16 +17164,19 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -19219,12 +19225,15 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc662_add_vol_ctl(spec, name, nid, i, 3); + index = 0; + } + err = __alc662_add_vol_ctl(spec, name, nid, index, 3); if (err < 0) return err; - err = __alc662_add_sw_ctl(spec, name, mix, i, 3); + err = __alc662_add_sw_ctl(spec, name, mix, index, 3); if (err < 0) return err; } -- cgit v1.1 From 15de7a41d30cfe8090efdc5fd6a92ed7a2d80ce7 Mon Sep 17 00:00:00 2001 From: Andres Salomon Date: Thu, 17 Feb 2011 19:07:17 -0800 Subject: mfd: mfd_cell is now implicitly available to wl1273 drivers The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. Signed-off-by: Andres Salomon Signed-off-by: Samuel Ortiz --- sound/soc/codecs/wl1273.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f..1ad0d5a 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -436,7 +436,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format); static int wl1273_probe(struct snd_soc_codec *codec) { - struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_core **core = + mfd_get_data(to_platform_device(codec->dev)); struct wl1273_priv *wl1273; int r; -- cgit v1.1 From 0638d56fbb6cf8367fcf01a1febf6a191b0e0704 Mon Sep 17 00:00:00 2001 From: Andres Salomon Date: Thu, 17 Feb 2011 19:07:20 -0800 Subject: mfd: mfd_cell is now implicitly available to twl4030 drivers The cell's platform_data is now accessed with a helper function; change clients to use that, and remove the now-unused data_size. Signed-off-by: Andres Salomon Acked-by: Peter Ujfalusi Signed-off-by: Samuel Ortiz --- sound/soc/codecs/twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b..8512800 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -732,7 +733,8 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; + struct twl4030_codec_audio_data *pdata = + mfd_get_data(to_platform_device(codec->dev)); unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -2297,7 +2299,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); -- cgit v1.1 From d57763370e1e12dd72e5a7bc6d6a7644e0003593 Mon Sep 17 00:00:00 2001 From: Andres Salomon Date: Mon, 28 Feb 2011 17:24:03 +0100 Subject: asoc: davinci_voicecodec: use mfd_data instead of driver_data Use mfd_data for passing information from mfd drivers to soc clients. The mfd_cell's driver_data field is being phased out. Clients that were using driver_data now access .mfd_data via mfd_get_data(). Signed-off-by: Andres Salomon Acked-by: Mark Brown Signed-off-by: Samuel Ortiz --- sound/soc/codecs/cq93vc.c | 3 ++- sound/soc/davinci/davinci-vcif.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 347a567..b8066ef 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); + struct davinci_vc *davinci_vc = + mfd_get_data(to_platform_device(codec->dev)); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9d2afcc..13e05a3 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { static int davinci_vcif_probe(struct platform_device *pdev) { - struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vc *davinci_vc = mfd_get_data(pdev); struct davinci_vcif_dev *davinci_vcif_dev; int ret; -- cgit v1.1 From dab1547a011b221308b6e991405677c78e1a8956 Mon Sep 17 00:00:00 2001 From: Andres Salomon Date: Thu, 17 Feb 2011 19:07:27 -0800 Subject: asoc: wm8400-codec: Use mfd_data instead of driver_data Use mfd_data for passing information from mfd drivers to soc clients. The mfd_cell's driver_data field is being phased out. Clients that were using driver_data now access .mfd_data via mfd_get_data(). Signed-off-by: Andres Salomon Acked-by: Mark Brown Signed-off-by: Samuel Ortiz --- sound/soc/codecs/wm8400.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 3c3bc07..736b785 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include @@ -1377,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work) static int wm8400_codec_probe(struct snd_soc_codec *codec) { - struct wm8400 *wm8400 = dev_get_platdata(codec->dev); + struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev)); struct wm8400_priv *priv; int ret; u16 reg; -- cgit v1.1 From b769f49463711205d57286e64cf535ed4daf59e9 Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Wed, 23 Mar 2011 10:53:41 -0400 Subject: sound/oss: remove offset from load_patch callbacks Was: [PATCH] sound/oss/midi_synth: prevent underflow, use of uninitialized value, and signedness issue The offset passed to midi_synth_load_patch() can be essentially arbitrary. If it's greater than the header length, this will result in a copy_from_user(dst, src, negative_val). While this will just return -EFAULT on x86, on other architectures this may cause memory corruption. Additionally, the length field of the sysex_info structure may not be initialized prior to its use. Finally, a signed comparison may result in an unintentionally large loop. On suggestion by Takashi Iwai, version two removes the offset argument from the load_patch callbacks entirely, which also resolves similar issues in opl3. Compile tested only. v3 adjusts comments and hopefully gets copy offsets right. Signed-off-by: Dan Rosenberg Signed-off-by: Takashi Iwai --- sound/oss/dev_table.h | 2 +- sound/oss/midi_synth.c | 30 +++++++++++++----------------- sound/oss/midi_synth.h | 2 +- sound/oss/opl3.c | 8 ++------ sound/oss/sequencer.c | 2 +- 5 files changed, 18 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h index b7617be..0199a31 100644 --- a/sound/oss/dev_table.h +++ b/sound/oss/dev_table.h @@ -271,7 +271,7 @@ struct synth_operations void (*reset) (int dev); void (*hw_control) (int dev, unsigned char *event); int (*load_patch) (int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void (*aftertouch) (int dev, int voice, int pressure); void (*controller) (int dev, int voice, int ctrl_num, int value); void (*panning) (int dev, int voice, int value); diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 3c09374..2292c23 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -476,7 +476,7 @@ EXPORT_SYMBOL(midi_synth_hw_control); int midi_synth_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { int orig_dev = synth_devs[dev]->midi_dev; @@ -491,33 +491,29 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, if (!prefix_cmd(orig_dev, 0xf0)) return 0; + /* Invalid patch format */ if (format != SYSEX_PATCH) - { -/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/ return -EINVAL; - } + + /* Patch header too short */ if (count < hdr_size) - { -/* printk("MIDI Error: Patch header too short\n");*/ return -EINVAL; - } + count -= hdr_size; /* - * Copy the header from user space but ignore the first bytes which have - * been transferred already. + * Copy the header from user space */ - if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs)) + if (copy_from_user(&sysex, addr, hdr_size)) return -EFAULT; - - if (count < sysex.len) - { -/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/ + + /* Sysex record too short */ + if ((unsigned)count < (unsigned)sysex.len) sysex.len = count; - } - left = sysex.len; - src_offs = 0; + + left = sysex.len; + src_offs = 0; for (i = 0; i < left && !signal_pending(current); i++) { diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h index 6bc9d00..b64ddd6 100644 --- a/sound/oss/midi_synth.h +++ b/sound/oss/midi_synth.h @@ -8,7 +8,7 @@ int midi_synth_open (int dev, int mode); void midi_synth_close (int dev); void midi_synth_hw_control (int dev, unsigned char *event); int midi_synth_load_patch (int dev, int format, const char __user * addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void midi_synth_panning (int dev, int channel, int pressure); void midi_synth_aftertouch (int dev, int channel, int pressure); void midi_synth_controller (int dev, int channel, int ctrl_num, int value); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 938c48c..cbf9574 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -820,7 +820,7 @@ static void opl3_hw_control(int dev, unsigned char *event) } static int opl3_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { struct sbi_instrument ins; @@ -830,11 +830,7 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, return -EINVAL; } - /* - * What the fuck is going on here? We leave junk in the beginning - * of ins and then check the field pretty close to that beginning? - */ - if(copy_from_user(&((char *) &ins)[offs], addr + offs, sizeof(ins) - offs)) + if (copy_from_user(&ins, addr, sizeof(ins))) return -EFAULT; if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 5ea1098..30bcfe4 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -241,7 +241,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun return -ENXIO; fmt = (*(short *) &event_rec[0]) & 0xffff; - err = synth_devs[dev]->load_patch(dev, fmt, buf, p + 4, c, 0); + err = synth_devs[dev]->load_patch(dev, fmt, buf + p, c, 0); if (err < 0) return err; -- cgit v1.1 From 4d00135a680727f6c3be78f8befaac009030e4df Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Wed, 23 Mar 2011 11:42:57 -0400 Subject: sound/oss/opl3: validate voice and channel indexes User-controllable indexes for voice and channel values may cause reading and writing beyond the bounds of their respective arrays, leading to potentially exploitable memory corruption. Validate these indexes. Signed-off-by: Dan Rosenberg Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/oss/opl3.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index cbf9574..407cd67 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -845,6 +845,10 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, static void opl3_panning(int dev, int voice, int value) { + + if (voice < 0 || voice >= devc->nr_voice) + return; + devc->voc[voice].panning = value; } @@ -1062,8 +1066,15 @@ static int opl3_alloc_voice(int dev, int chn, int note, struct voice_alloc_info static void opl3_setup_voice(int dev, int voice, int chn) { - struct channel_info *info = - &synth_devs[dev]->chn_info[chn]; + struct channel_info *info; + + if (voice < 0 || voice >= devc->nr_voice) + return; + + if (chn < 0 || chn > 15) + return; + + info = &synth_devs[dev]->chn_info[chn]; opl3_set_instr(dev, voice, info->pgm_num); -- cgit v1.1 From e19869204fca4ae28b6e4d8f5e20849e9f7b18bd Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 17 Feb 2011 14:26:51 +0100 Subject: ALSA: usb-audio: add Cakewalk UM-1G support Add a quirk for the Cakewalk UM-1G USB MIDI interface in "advanced driver" mode. (It already works in standard mode.) Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c0dcfca..196c753 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1568,6 +1568,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0582, 0x0104), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UM-1G", */ + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + } +}, +{ /* has ID 0x0110 when not in Advanced Driver mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.1 From cb6f4b55f5907528d8a1a927b850c9eb04d4ef90 Mon Sep 17 00:00:00 2001 From: "Keith A. Milner" Date: Mon, 21 Mar 2011 20:15:08 +0000 Subject: ALSA: usb-audio - Support for Boss JS-8 Jam Station Signed-off-by: Keith A. Milner Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 196c753..c66d3f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1581,6 +1581,33 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Boss JS-8 Jam Station */ + USB_DEVICE(0x0582, 0x0109), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "JS-8", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ /* has ID 0x0110 when not in Advanced Driver mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.1 From 20b67dddcc5f29d3d0c900225d85e0ac655bc69d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Mar 2011 22:54:32 +0100 Subject: ALSA: hda - Fix SPDIF out regression on ALC889 The commit 5a8cfb4e8ae317d283f84122ed20faa069c5e0c4 ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization changed to use the default initialization method for ALC889, but this caused a regression on SPDIF output on some machines. This seems due to the COEF setup included in the default init procedure. For making SPDIF working again, the COEF-setup has to be avoided for the id 0889. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=24342 Cc: 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 5d582de..0ef0035 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1290,7 +1290,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0883: case 0x10ec0885: case 0x10ec0887: - case 0x10ec0889: + /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ alc889_coef_init(codec); break; case 0x10ec0888: -- cgit v1.1 From 3674f19dabd15f9541079a588149a370d888f4e6 Mon Sep 17 00:00:00 2001 From: Benjamin Herrenschmidt Date: Fri, 25 Mar 2011 17:51:54 +1100 Subject: ALSA: vmalloc buffers should use normal mmap It's a big no-no to use pgprot_noncached() when mmap'ing such buffers into userspace since they are mapped cachable in kernel space. This can cause all sort of interesting things ranging from to garbled sound to lockups on various architectures. I've observed that usb-audio is broken on powerpc 4xx for example because of that. Also remove the now unused snd_pcm_lib_mmap_noncached(). It's an arch business to know when to use uncached mappings, there's already hacks for MIPS inside snd_pcm_default_mmap() and other archs are supposed to use dma_mmap_coherent(). (See my separate patch that adds dma_mmap_coherent() to powerpc) Signed-off-by: Benjamin Herrenschmidt CC: Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ae42b65..fe5c803 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3201,15 +3201,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ -/* mmap callback with pgprot_noncached */ -int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); - return snd_pcm_default_mmap(substream, area); -} -EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); - /* * mmap DMA buffer */ -- cgit v1.1 From 677cd904aba939bc4cfdc3c1eada8ec46582127e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 7 Feb 2011 15:19:34 +0100 Subject: ALSA: HDA: New AD1984A model for Dell Precision R5500 For codec AD1984A, add a new model to support Dell Precision R5500 or the microphone jack won't work correctly. BugLink: http://bugs.launchpad.net/bugs/741516 Tested-by: Kent Baxley Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 89 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 89 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 734c6ee..2942d2a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4256,6 +4256,84 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * Precision R5500 + * 0x12 - HP/line-out + * 0x13 - speaker (mono) + * 0x15 - mic-in + */ + +static struct hda_verb ad1984a_precision_verbs[] = { + /* Unmute main output path */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */ + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Select mic as input */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */ + /* Configure as mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ + /* HP unmute */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* turn on EAPD */ + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* unsolicited event for pin-sense */ + {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_precision_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { } /* end */ +}; + + +/* mute internal speaker if HP is plugged */ +static void ad1984a_precision_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x12); + snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_precision_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1984a_precision_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1984a_precision_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1984a_precision_automute(codec); + return 0; +} + + +/* * HP Touchsmart * port-A (0x11) - front hp-out * port-B (0x14) - unused @@ -4384,6 +4462,7 @@ enum { AD1884A_MOBILE, AD1884A_THINKPAD, AD1984A_TOUCHSMART, + AD1984A_PRECISION, AD1884A_MODELS }; @@ -4393,9 +4472,11 @@ static const char * const ad1884a_models[AD1884A_MODELS] = { [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", [AD1984A_TOUCHSMART] = "touchsmart", + [AD1984A_PRECISION] = "precision", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { + SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION), SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), @@ -4489,6 +4570,14 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_PRECISION: + spec->mixers[0] = ad1984a_precision_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1984a_precision_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_precision_unsol_event; + codec->patch_ops.init = ad1984a_precision_init; + break; case AD1984A_TOUCHSMART: spec->mixers[0] = ad1984a_touchsmart_mixers; spec->init_verbs[0] = ad1984a_touchsmart_verbs; -- cgit v1.1 From 7bf76c33e9a1ecb2a15f1a066d4e032b5d0922a7 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Fri, 25 Mar 2011 15:25:46 +1300 Subject: ALSA: asihpi - Support single-rate no-SRC cards Cards without settable local samplerate and without SRC still must have a valid samplerate. This fixed rate is determined by reading the current rate for the card. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 0ac1f98..22606e3 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -381,13 +381,13 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, "No local sampleclock, err %d\n", err); } - for (idx = 0; idx < 100; idx++) { - if (hpi_sample_clock_query_local_rate( - h_control, idx, &sample_rate)) { - if (!idx) - snd_printk(KERN_ERR - "Local rate query failed\n"); - + for (idx = -1; idx < 100; idx++) { + if (idx == -1) { + if (hpi_sample_clock_get_sample_rate(h_control, + &sample_rate)) + continue; + } else if (hpi_sample_clock_query_local_rate(h_control, + idx, &sample_rate)) { break; } @@ -440,8 +440,6 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, } } - /* printk(KERN_INFO "Supported rates %X %d %d\n", - rates, rate_min, rate_max); */ pcmhw->rates = rates; pcmhw->rate_min = rate_min; pcmhw->rate_max = rate_max; -- cgit v1.1 From 26aebef420f8036213419b8a46e3a07db51439cd Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Fri, 25 Mar 2011 15:25:47 +1300 Subject: ALSA: asihpi - Improve non-busmaster adapter operation Make playback silence callback a no-op, card automatically outputs silence when written data runs out. Increasing update interval and thus minimum period avoids xrun on startup or because of timer jitter. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 22606e3..c90d77a 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1012,6 +1012,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames * 2, UINT_MAX); @@ -1054,7 +1055,7 @@ static int snd_card_asihpi_playback_copy(struct snd_pcm_substream *substream, hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream, runtime->dma_area, len, &dpcm->format)); - dpcm->pcm_buf_host_rw_ofs = dpcm->pcm_buf_host_rw_ofs + len; + dpcm->pcm_buf_host_rw_ofs += len; return 0; } @@ -1064,16 +1065,11 @@ static int snd_card_asihpi_playback_silence(struct snd_pcm_substream * snd_pcm_uframes_t pos, snd_pcm_uframes_t count) { - unsigned int len; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - - len = frames_to_bytes(runtime, count); - VPRINTK1(KERN_INFO "playback silence %u bytes\n", len); - - memset(runtime->dma_area, 0, len); - hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream, - runtime->dma_area, len, &dpcm->format)); + /* Usually writes silence to DMA buffer, which should be overwritten + by real audio later. Our fifos cannot be overwritten, and are not + free-running DMAs. Silence is output on fifo underflow. + This callback is still required to allow the copy callback to be used. + */ return 0; } @@ -2885,6 +2881,9 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, if (err) asihpi->update_interval_frames = 512; + if (!asihpi->support_mmap) + asihpi->update_interval_frames *= 2; + hpi_handle_error(hpi_instream_open(asihpi->adapter_index, 0, &h_stream)); -- cgit v1.1 From b2e65c8e9133218eb28c30e79ddd3d66d4666ba0 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Fri, 25 Mar 2011 15:25:48 +1300 Subject: ALSA: asihpi - Update verbose debug print macros Replace local VPRINTK1 with snd_printdd. Create local snd_printddd instead of VPRINTK2 for most verbose debug. In most cases let snd_printk supply default level for messages. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 100 ++++++++++++++++++++++------------------------ 1 file changed, 47 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index c90d77a..f53a31e 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -22,21 +22,6 @@ * for any purpose including commercial applications. */ -/* >0: print Hw params, timer vars. >1: print stream write/copy sizes */ -#define REALLY_VERBOSE_LOGGING 0 - -#if REALLY_VERBOSE_LOGGING -#define VPRINTK1 snd_printd -#else -#define VPRINTK1(...) -#endif - -#if REALLY_VERBOSE_LOGGING > 1 -#define VPRINTK2 snd_printd -#else -#define VPRINTK2(...) -#endif - #include "hpi_internal.h" #include "hpimsginit.h" #include "hpioctl.h" @@ -57,11 +42,25 @@ #include #include - MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. "); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); +#if defined CONFIG_SND_DEBUG_VERBOSE +/** + * snd_printddd - very verbose debug printk + * @format: format string + * + * Works like snd_printk() for debugging purposes. + * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. + * Must set snd module debug parameter to 3 to enable at runtime. + */ +#define snd_printddd(format, args...) \ + __snd_printk(3, __FILE__, __LINE__, format, ##args) +#else +#define snd_printddd(format, args...) do { } while (0) +#endif + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; @@ -289,7 +288,6 @@ static u16 handle_error(u16 err, int line, char *filename) #define hpi_handle_error(x) handle_error(x, __LINE__, __FILE__) /***************************** GENERAL PCM ****************/ -#if REALLY_VERBOSE_LOGGING static void print_hwparams(struct snd_pcm_hw_params *p) { snd_printd("HWPARAMS \n"); @@ -304,9 +302,6 @@ static void print_hwparams(struct snd_pcm_hw_params *p) snd_printd("periods %d \n", params_periods(p)); snd_printd("buffer_size %d \n", params_buffer_size(p)); } -#else -#define print_hwparams(x) -#endif static snd_pcm_format_t hpi_to_alsa_formats[] = { -1, /* INVALID */ @@ -464,7 +459,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, if (err) return err; - VPRINTK1(KERN_INFO "format %d, %d chans, %d_hz\n", + snd_printdd("format %d, %d chans, %d_hz\n", format, params_channels(params), params_rate(params)); @@ -487,13 +482,12 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, err = hpi_stream_host_buffer_attach(dpcm->h_stream, params_buffer_bytes(params), runtime->dma_addr); if (err == 0) { - VPRINTK1(KERN_INFO + snd_printdd( "stream_host_buffer_attach succeeded %u %lu\n", params_buffer_bytes(params), (unsigned long)runtime->dma_addr); } else { - snd_printd(KERN_INFO - "stream_host_buffer_attach error %d\n", + snd_printd("stream_host_buffer_attach error %d\n", err); return -ENOMEM; } @@ -502,7 +496,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, &dpcm->hpi_buffer_attached, NULL, NULL, NULL); - VPRINTK1(KERN_INFO "stream_host_buffer_attach status 0x%x\n", + snd_printdd("stream_host_buffer_attach status 0x%x\n", dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); @@ -515,7 +509,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->bytes_per_sec = bytes_per_sec; dpcm->buffer_bytes = params_buffer_bytes(params); dpcm->period_bytes = params_period_bytes(params); - VPRINTK1(KERN_INFO "buffer_bytes=%d, period_bytes=%d, bps=%d\n", + snd_printdd("buffer_bytes=%d, period_bytes=%d, bps=%d\n", dpcm->buffer_bytes, dpcm->period_bytes, bytes_per_sec); return 0; @@ -571,7 +565,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, struct snd_pcm_substream *s; u16 e; - VPRINTK1(KERN_INFO "%c%d trigger\n", + snd_printdd("%c%d trigger\n", SCHR(substream->stream), substream->number); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -595,7 +589,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, * data?? */ unsigned int preload = ds->period_bytes * 1; - VPRINTK2(KERN_INFO "%d preload x%x\n", s->number, preload); + snd_printddd("%d preload x%x\n", s->number, preload); hpi_handle_error(hpi_outstream_write_buf( ds->h_stream, &runtime->dma_area[0], @@ -605,7 +599,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } if (card->support_grouping) { - VPRINTK1(KERN_INFO "\t%c%d group\n", + snd_printdd("\t%c%d group\n", SCHR(s->stream), s->number); e = hpi_stream_group_add( @@ -620,7 +614,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } else break; } - VPRINTK1(KERN_INFO "start\n"); + snd_printdd("start\n"); /* start the master stream */ snd_card_asihpi_pcm_timer_start(substream); if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) || @@ -642,14 +636,14 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, s->runtime->status->state = SNDRV_PCM_STATE_SETUP; if (card->support_grouping) { - VPRINTK1(KERN_INFO "\t%c%d group\n", + snd_printdd("\t%c%d group\n", SCHR(s->stream), s->number); snd_pcm_trigger_done(s, substream); } else break; } - VPRINTK1(KERN_INFO "stop\n"); + snd_printdd("stop\n"); /* _prepare and _hwparams reset the stream */ hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); @@ -662,12 +656,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - VPRINTK1(KERN_INFO "pause release\n"); + snd_printdd("pause release\n"); hpi_handle_error(hpi_stream_start(dpcm->h_stream)); snd_card_asihpi_pcm_timer_start(substream); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - VPRINTK1(KERN_INFO "pause\n"); + snd_printdd("pause\n"); snd_card_asihpi_pcm_timer_stop(substream); hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); break; @@ -739,7 +733,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) u16 state; u32 buffer_size, bytes_avail, samples_played, on_card_bytes; - VPRINTK1(KERN_INFO "%c%d snd_card_asihpi_timer_function\n", + snd_printdd("%c%d snd_card_asihpi_timer_function\n", SCHR(substream->stream), substream->number); /* find minimum newdata and buffer pos in group */ @@ -768,10 +762,10 @@ static void snd_card_asihpi_timer_function(unsigned long data) if ((bytes_avail == 0) && (on_card_bytes < ds->pcm_buf_host_rw_ofs)) { hpi_handle_error(hpi_stream_start(ds->h_stream)); - VPRINTK1(KERN_INFO "P%d start\n", s->number); + snd_printdd("P%d start\n", s->number); } } else if (state == HPI_STATE_DRAINED) { - VPRINTK1(KERN_WARNING "P%d drained\n", + snd_printd(KERN_WARNING "P%d drained\n", s->number); /*snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); continue; */ @@ -792,13 +786,13 @@ static void snd_card_asihpi_timer_function(unsigned long data) newdata); } - VPRINTK1(KERN_INFO "PB timer hw_ptr x%04lX, appl_ptr x%04lX\n", + snd_printdd("hw_ptr x%04lX, appl_ptr x%04lX\n", (unsigned long)frames_to_bytes(runtime, runtime->status->hw_ptr), (unsigned long)frames_to_bytes(runtime, runtime->control->appl_ptr)); - VPRINTK1(KERN_INFO "%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X," + snd_printdd("%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X," " aux=x%04X space=x%04X\n", loops, SCHR(s->stream), s->number, state, ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, (int)bytes_avail, @@ -820,7 +814,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) next_jiffies = max(next_jiffies, 1U); dpcm->timer.expires = jiffies + next_jiffies; - VPRINTK1(KERN_INFO "jif %d buf pos x%04X newdata x%04X xfer x%04X\n", + snd_printdd("jif %d buf pos x%04X newdata x%04X xfer x%04X\n", next_jiffies, pcm_buf_dma_ofs, newdata, xfercount); snd_pcm_group_for_each_entry(s, substream) { @@ -835,7 +829,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) if (xfercount && (on_card_bytes <= ds->period_bytes)) { if (card->support_mmap) { if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - VPRINTK2(KERN_INFO "P%d write x%04x\n", + snd_printddd("P%d write x%04x\n", s->number, ds->period_bytes); hpi_handle_error( @@ -846,7 +840,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) xfercount, &ds->format)); } else { - VPRINTK2(KERN_INFO "C%d read x%04x\n", + snd_printddd("C%d read x%04x\n", s->number, xfercount); hpi_handle_error( @@ -869,7 +863,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - /* snd_printd(KERN_INFO "Playback ioctl %d\n", cmd); */ + snd_printdd(KERN_INFO "Playback ioctl %d\n", cmd); return snd_pcm_lib_ioctl(substream, cmd, arg); } @@ -879,7 +873,7 @@ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream * struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - VPRINTK1(KERN_INFO "playback prepare %d\n", substream->number); + snd_printdd("playback prepare %d\n", substream->number); hpi_handle_error(hpi_outstream_reset(dpcm->h_stream)); dpcm->pcm_buf_host_rw_ofs = 0; @@ -896,7 +890,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t ptr; ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes); - /* VPRINTK2(KERN_INFO "playback_pointer=x%04lx\n", (unsigned long)ptr); */ + snd_printddd("playback_pointer=x%04lx\n", (unsigned long)ptr); return ptr; } @@ -1018,7 +1012,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); - VPRINTK1(KERN_INFO "playback open\n"); + snd_printdd("playback open\n"); return 0; } @@ -1029,7 +1023,7 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream) struct snd_card_asihpi_pcm *dpcm = runtime->private_data; hpi_handle_error(hpi_outstream_close(dpcm->h_stream)); - VPRINTK1(KERN_INFO "playback close\n"); + snd_printdd("playback close\n"); return 0; } @@ -1049,7 +1043,7 @@ static int snd_card_asihpi_playback_copy(struct snd_pcm_substream *substream, if (copy_from_user(runtime->dma_area, src, len)) return -EFAULT; - VPRINTK2(KERN_DEBUG "playback copy%d %u bytes\n", + snd_printddd("playback copy%d %u bytes\n", substream->number, len); hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream, @@ -1104,7 +1098,7 @@ snd_card_asihpi_capture_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - VPRINTK2(KERN_INFO "capture pointer %d=%d\n", + snd_printddd("capture pointer %d=%d\n", substream->number, dpcm->pcm_buf_dma_ofs); /* NOTE Unlike playback can't use actual samples_played for the capture position, because those samples aren't yet in @@ -1129,7 +1123,7 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) dpcm->pcm_buf_dma_ofs = 0; dpcm->pcm_buf_elapsed_dma_ofs = 0; - VPRINTK1("Capture Prepare %d\n", substream->number); + snd_printdd("Capture Prepare %d\n", substream->number); return 0; } @@ -1192,7 +1186,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (dpcm == NULL) return -ENOMEM; - VPRINTK1("hpi_instream_open adapter %d stream %d\n", + snd_printdd("capture open adapter %d stream %d\n", card->adapter_index, substream->number); err = hpi_handle_error( @@ -1262,7 +1256,7 @@ static int snd_card_asihpi_capture_copy(struct snd_pcm_substream *substream, len = frames_to_bytes(runtime, count); - VPRINTK2(KERN_INFO "capture copy%d %d bytes\n", substream->number, len); + snd_printddd("capture copy%d %d bytes\n", substream->number, len); hpi_handle_error(hpi_instream_read_buf(dpcm->h_stream, runtime->dma_area, len)); @@ -2906,7 +2900,6 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->support_mrx ); - err = snd_card_asihpi_pcm_new(asihpi, 0, pcm_substreams); if (err < 0) { snd_printk(KERN_ERR "pcm_new failed\n"); @@ -2941,6 +2934,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, sprintf(card->longname, "%s %i", card->shortname, asihpi->adapter_index); err = snd_card_register(card); + if (!err) { hpi_card->snd_card_asihpi = card; dev++; -- cgit v1.1 From a45e3d6b13e97506b616980c0f122c3389bcefa4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Mar 2011 09:50:15 +0100 Subject: ALSA: Fix yet another race in disconnection This patch fixes a race between snd_card_file_remove() and snd_card_disconnect(). When the card is added to shutdown_files list in snd_card_disconnect(), but it's freed in snd_card_file_remove() at the same time, the shutdown_files list gets corrupted. The list member must be freed in snd_card_file_remove() as well. Reported-and-tested-by: Russ Dill Cc: Signed-off-by: Takashi Iwai --- sound/core/init.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 3e65da2..a0080aa 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -848,6 +848,7 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return -ENOMEM; mfile->file = file; mfile->disconnected_f_op = NULL; + INIT_LIST_HEAD(&mfile->shutdown_list); spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); @@ -883,6 +884,9 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) list_for_each_entry(mfile, &card->files_list, list) { if (mfile->file == file) { list_del(&mfile->list); + spin_lock(&shutdown_lock); + list_del(&mfile->shutdown_list); + spin_unlock(&shutdown_lock); if (mfile->disconnected_f_op) fops_put(mfile->disconnected_f_op); found = mfile; -- cgit v1.1 From c6b358748e19ce7e230b0926ac42696bc485a562 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Mar 2011 12:05:31 +0200 Subject: ALSA: hda - Fix pin-config of Gigabyte mobo Use pin-fix instead of the static quirk for Gigabyte mobos 1458:a002. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=677256 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ef0035..12c6f45 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9863,7 +9863,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), @@ -10700,6 +10699,7 @@ enum { PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, + PINFIX_GIGABYTE_880GM, }; static const struct alc_fixup alc882_fixups[] = { @@ -10731,6 +10731,13 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_SKU, .v.sku = ALC_FIXUP_SKU_IGNORE, }, + [PINFIX_GIGABYTE_880GM] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x1114410 }, /* set as speaker */ + { } + } + }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -10738,6 +10745,7 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM), {} }; @@ -18774,8 +18782,6 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", - ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), @@ -19449,6 +19455,7 @@ enum { ALC662_FIXUP_IDEAPAD, ALC272_FIXUP_MARIO, ALC662_FIXUP_CZC_P10T, + ALC662_FIXUP_GIGABYTE, }; static const struct alc_fixup alc662_fixups[] = { @@ -19477,12 +19484,20 @@ static const struct alc_fixup alc662_fixups[] = { {} } }, + [ALC662_FIXUP_GIGABYTE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x1114410 }, /* set as speaker */ + { } + } + }, }; static struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), -- cgit v1.1