From 2fbe74b90bafebce615466b4c20f96b0465df1ae Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 16 Dec 2009 16:54:43 +0100 Subject: [PATCH 01/11] sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot() limit and jiffies are unsigned so the test did not work. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/pss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 83f5ee236b1..e19dd5dcc2d 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc) unsigned long i, limit = jiffies + HZ/10; outw(0x2000, REG(PSS_CONTROL)); - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) inw(REG(PSS_CONTROL)); outw(0x0000, REG(PSS_CONTROL)); return 1; @@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size outw(0, REG(PSS_DATA)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit - jiffies >= 0); i++) + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) val = inw(REG(PSS_STATUS)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) { val = inw(REG(PSS_STATUS)); if (val & 0x4000) From 3e85fd614c7b6bb7f33bb04a0dcb5a3bfca4c0fe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:27:24 +0100 Subject: [PATCH 02/11] sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM buffer When allocating the PCM buffer, use vmalloc_user() instead of vmalloc(). Otherwise, it would be possible for applications to play the previous contents of the kernel memory to the speakers, or to read it directly if the buffer is exported to userspace. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 2 +- sound/usb/usbaudio.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 8691f4cf619..f1d9d16b548 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -609,7 +609,7 @@ static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, /* alloc virtual 'dma' area */ if (runtime->dma_area) vfree(runtime->dma_area); - runtime->dma_area = vmalloc(size); + runtime->dma_area = vmalloc_user(size); if (runtime->dma_area == NULL) return -ENOMEM; runtime->dma_bytes = size; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index d057e648964..5cfa608823f 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -51,7 +51,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s return 0; /* already enough large */ vfree(runtime->dma_area); } - runtime->dma_area = vmalloc_32(size); + runtime->dma_area = vmalloc_32_user(size); if (! runtime->dma_area) return -ENOMEM; runtime->dma_bytes = size; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b074a594c59..4963defee18 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -752,7 +752,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s return 0; /* already large enough */ vfree(runtime->dma_area); } - runtime->dma_area = vmalloc(size); + runtime->dma_area = vmalloc_user(size); if (!runtime->dma_area) return -ENOMEM; runtime->dma_bytes = size; From d49464318a7c51676c44cbd1e2480f651cc43807 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 20:25:30 +0100 Subject: [PATCH 03/11] ALSA: aaci - Fix a typo Fixed a typo of the max buffer size specified for buffer allocation changed in the commit d6797322231af98b9bb4afb175dd614cf511e5f7. Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1497dce1b04..ae38f2c342c 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1028,7 +1028,7 @@ static int __devinit aaci_init_pcm(struct aaci *aaci) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - NULL, 0, 64 * 104); + NULL, 0, 64 * 1024); } return ret; From 6ca867c827c84d21316e9dc4035abe9480f8347c Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:35 +0000 Subject: [PATCH 04/11] ALSA: AACI: simplify codec rate information There's no need for a specific rule; ALSA's generic AC'97 support calculates the necessary rate constraint information itself, and we can use this directly. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 75 ++---------------------------------------------- 1 file changed, 3 insertions(+), 72 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index ae38f2c342c..ea3be874c84 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -330,63 +330,6 @@ static irqreturn_t aaci_irq(int irq, void *devid) /* * ALSA support. */ - -struct aaci_stream { - unsigned char codec_idx; - unsigned char rate_idx; -}; - -static struct aaci_stream aaci_streams[] = { - [ACSTREAM_FRONT] = { - .codec_idx = 0, - .rate_idx = AC97_RATES_FRONT_DAC, - }, - [ACSTREAM_SURROUND] = { - .codec_idx = 0, - .rate_idx = AC97_RATES_SURR_DAC, - }, - [ACSTREAM_LFE] = { - .codec_idx = 0, - .rate_idx = AC97_RATES_LFE_DAC, - }, -}; - -static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid) -{ - struct aaci_stream *s = aaci_streams + streamid; - return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx]; -} - -static unsigned int rate_list[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 -}; - -/* - * Double-rate rule: we can support double rate iff channels == 2 - * (unimplemented) - */ -static int -aaci_rule_rate_by_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule) -{ - struct aaci *aaci = rule->private; - unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512; - struct snd_interval *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS); - - switch (c->max) { - case 6: - rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE); - case 4: - rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND); - case 2: - rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT); - } - - return snd_interval_list(hw_param_interval(p, rule->var), - ARRAY_SIZE(rate_list), rate_list, - rate_mask); -} - static struct snd_pcm_hardware aaci_hw_info = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -400,10 +343,7 @@ static struct snd_pcm_hardware aaci_hw_info = { */ .formats = SNDRV_PCM_FMTBIT_S16_LE, - /* should this be continuous or knot? */ - .rates = SNDRV_PCM_RATE_CONTINUOUS, - .rate_max = 48000, - .rate_min = 4000, + /* rates are setup from the AC'97 codec */ .channels_min = 2, .channels_max = 6, .buffer_bytes_max = 64 * 1024, @@ -423,6 +363,8 @@ static int __aaci_pcm_open(struct aaci *aaci, aacirun->substream = substream; runtime->private_data = aacirun; runtime->hw = aaci_hw_info; + runtime->hw.rates = aacirun->pcm->rates; + snd_pcm_limit_hw_rates(runtime); /* * FIXME: ALSA specifies fifo_size in bytes. If we're in normal @@ -433,17 +375,6 @@ static int __aaci_pcm_open(struct aaci *aaci, */ runtime->hw.fifo_size = aaci->fifosize * 2; - /* - * Add rule describing hardware rate dependency - * on the number of channels. - */ - ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - aaci_rule_rate_by_channels, aaci, - SNDRV_PCM_HW_PARAM_CHANNELS, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (ret) - goto out; - ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED, DRIVER_NAME, aaci); if (ret) From 4e30b69108b20eca80f1a323f969bf7629c7795f Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:37 +0000 Subject: [PATCH 05/11] ALSA: AACI: cleanup aaci_pcm_hw_params Since the recording and playback paths are now the same, eliminate the needless conditionals. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index ea3be874c84..2e28748a3d8 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -438,18 +438,14 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (err < 0) - goto out; + if (err >= 0) { + err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + params_channels(params), + aacirun->pcm->r[0].slots); - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - if (err) - goto out; + aacirun->pcm_open = err == 0; + } - aacirun->pcm_open = 1; - - out: return err; } @@ -458,7 +454,7 @@ 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->start = (void *)runtime->dma_area; + aacirun->start = runtime->dma_area; aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = From d3aee7996c30f928bbbbfd0994148e35d2e83084 Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:40 +0000 Subject: [PATCH 06/11] ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 2e28748a3d8..b88bbded2f4 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -444,6 +444,11 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aacirun->pcm->r[0].slots); aacirun->pcm_open = err == 0; + aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; + aacirun->fifosz = aaci->fifosize * 4; + + if (aacirun->cr & CR_COMPACT) + aacirun->fifosz >>= 1; } return err; @@ -554,14 +559,9 @@ static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, * Enable FIFO, compact mode, 16 bits per sample. * FIXME: double rate slots? */ - if (ret >= 0) { - aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; + if (ret >= 0) aacirun->cr |= channels_to_txmask[channels]; - aacirun->fifosz = aaci->fifosize * 4; - if (aacirun->cr & CR_COMPACT) - aacirun->fifosz >>= 1; - } return ret; } @@ -648,18 +648,10 @@ static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, int ret; ret = aaci_pcm_hw_params(substream, aacirun, params); - - if (ret >= 0) { - aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; - + if (ret >= 0) /* Line in record: slot 3 and 4 */ aacirun->cr |= CR_SL3 | CR_SL4; - aacirun->fifosz = aaci->fifosize * 4; - - if (aacirun->cr & CR_COMPACT) - aacirun->fifosz >>= 1; - } return ret; } From a08d56583f6b87e2981d1b6e9ee891bdc741cc44 Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:45 +0000 Subject: [PATCH 07/11] ALSA: AACI: add double-rate support Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index b88bbded2f4..b377370af2d 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -366,6 +366,10 @@ static int __aaci_pcm_open(struct aaci *aaci, 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); + /* * FIXME: ALSA specifies fifo_size in bytes. If we're in normal * mode, each 32-bit word contains one sample. If we're in @@ -439,9 +443,12 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (err >= 0) { - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + unsigned int rate = params_rate(params); + int dbl = rate > 48000; + + err = snd_ac97_pcm_open(aacirun->pcm, rate, params_channels(params), - aacirun->pcm->r[0].slots); + aacirun->pcm->r[dbl].slots); aacirun->pcm_open = err == 0; aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; @@ -808,6 +815,12 @@ static struct ac97_pcm ac97_defs[] __devinitdata = { (1 << AC97_SLOT_PCM_SRIGHT) | (1 << AC97_SLOT_LFE), }, + [1] = { + .slots = (1 << AC97_SLOT_PCM_LEFT) | + (1 << AC97_SLOT_PCM_RIGHT) | + (1 << AC97_SLOT_PCM_LEFT_0) | + (1 << AC97_SLOT_PCM_RIGHT_0), + }, }, }, [1] = { /* PCM in */ From d6a89fefa50feda5516cd5210ad0008a44632b52 Mon Sep 17 00:00:00 2001 From: Russell King Date: Fri, 18 Dec 2009 17:48:50 +0000 Subject: [PATCH 08/11] ALSA: AACI: switch to per-pcm locking We can use finer-grained locking, which makes things easier when we gain DMA support. Signed-off-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 49 ++++++++++++++++++++++++++++-------------------- sound/arm/aaci.h | 2 +- 2 files changed, 30 insertions(+), 21 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index b377370af2d..c5699863643 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -172,14 +172,15 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) return v; } -static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun) +static inline void +aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask) { u32 val; int timeout = 5000; do { val = readl(aacirun->base + AACI_SR); - } while (val & (SR_TXB|SR_RXB) && timeout--); + } while (val & mask && timeout--); } @@ -208,8 +209,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) writel(0, aacirun->base + AACI_IE); return; } - ptr = aacirun->ptr; + spin_lock(&aacirun->lock); + + ptr = aacirun->ptr; do { unsigned int len = aacirun->fifosz; u32 val; @@ -217,9 +220,9 @@ 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(&aaci->lock); + spin_unlock(&aacirun->lock); snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aaci->lock); + spin_lock(&aacirun->lock); } if (!(aacirun->cr & CR_EN)) break; @@ -245,7 +248,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) ptr = aacirun->start; } } while(1); + aacirun->ptr = ptr; + + spin_unlock(&aacirun->lock); } if (mask & ISR_URINTR) { @@ -263,6 +269,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) return; } + spin_lock(&aacirun->lock); + ptr = aacirun->ptr; do { unsigned int len = aacirun->fifosz; @@ -271,9 +279,9 @@ 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(&aaci->lock); + spin_unlock(&aacirun->lock); snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aaci->lock); + spin_lock(&aacirun->lock); } if (!(aacirun->cr & CR_EN)) break; @@ -301,6 +309,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) } while (1); aacirun->ptr = ptr; + + spin_unlock(&aacirun->lock); } } @@ -310,7 +320,6 @@ static irqreturn_t aaci_irq(int irq, void *devid) u32 mask; int i; - spin_lock(&aaci->lock); mask = readl(aaci->base + AACI_ALLINTS); if (mask) { u32 m = mask; @@ -320,7 +329,6 @@ static irqreturn_t aaci_irq(int irq, void *devid) } } } - spin_unlock(&aaci->lock); return mask ? IRQ_HANDLED : IRQ_NONE; } @@ -580,7 +588,7 @@ static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun) ie &= ~(IE_URIE|IE_TXIE); writel(ie, aacirun->base + AACI_IE); aacirun->cr &= ~CR_EN; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_TXB); writel(aacirun->cr, aacirun->base + AACI_TXCR); } @@ -588,7 +596,7 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun) { u32 ie; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_TXB); aacirun->cr |= CR_EN; ie = readl(aacirun->base + AACI_IE); @@ -599,12 +607,12 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun) static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned long flags; int ret = 0; - spin_lock_irqsave(&aaci->lock, flags); + spin_lock_irqsave(&aacirun->lock, flags); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: aaci_pcm_playback_start(aacirun); @@ -631,7 +639,8 @@ static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cm default: ret = -EINVAL; } - spin_unlock_irqrestore(&aaci->lock, flags); + + spin_unlock_irqrestore(&aacirun->lock, flags); return ret; } @@ -666,7 +675,7 @@ static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun) { u32 ie; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_RXB); ie = readl(aacirun->base + AACI_IE); ie &= ~(IE_ORIE | IE_RXIE); @@ -681,7 +690,7 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun) { u32 ie; - aaci_chan_wait_ready(aacirun); + aaci_chan_wait_ready(aacirun, SR_RXB); #ifdef DEBUG /* RX Timeout value: bits 28:17 in RXCR */ @@ -698,12 +707,11 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun) static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned long flags; int ret = 0; - spin_lock_irqsave(&aaci->lock, flags); + spin_lock_irqsave(&aacirun->lock, flags); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -732,7 +740,7 @@ static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd ret = -EINVAL; } - spin_unlock_irqrestore(&aaci->lock, flags); + spin_unlock_irqrestore(&aacirun->lock, flags); return ret; } @@ -933,7 +941,6 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) aaci = card->private_data; mutex_init(&aaci->ac97_sem); - spin_lock_init(&aaci->lock); aaci->card = card; aaci->dev = dev; @@ -1020,12 +1027,14 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) /* * Playback uses AACI channel 0 */ + spin_lock_init(&aaci->playback.lock); aaci->playback.base = aaci->base + AACI_CSCH1; aaci->playback.fifo = aaci->base + AACI_DR1; /* * Capture uses AACI channel 0 */ + spin_lock_init(&aaci->capture.lock); aaci->capture.base = aaci->base + AACI_CSCH1; aaci->capture.fifo = aaci->base + AACI_DR1; diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 924f69c1c44..6a4a2eebdda 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -202,6 +202,7 @@ struct aaci_runtime { void __iomem *base; void __iomem *fifo; + spinlock_t lock; struct ac97_pcm *pcm; int pcm_open; @@ -232,7 +233,6 @@ struct aaci { struct snd_ac97 *ac97; u32 maincr; - spinlock_t lock; struct aaci_runtime playback; struct aaci_runtime capture; From ef86f581f7e8b29cb58d7f4e892e1a91b3805124 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 19 Dec 2009 08:18:03 +0100 Subject: [PATCH 09/11] ALSA: Use kzalloc for allocating only one thing Use kzalloc rather than kcalloc(1,...) The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // @@ @@ - kcalloc(1, + kzalloc( ...) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_midi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index cb9aa4c4edd..4be562b2cf2 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *card, int device) err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); if (err < 0) return err; - mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + mpu = kzalloc(sizeof(*mpu), GFP_KERNEL); if (mpu == NULL) { snd_device_free(card, rmidi); return -ENOMEM; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aeed4cc5aa7..20c1828e4ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12857,7 +12857,7 @@ static int patch_alc268(struct hda_codec *codec) int board_config; int i, has_beep, err; - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; From 40962d7c741de1c21b6ce8516c1d9f8836fb383e Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 19 Dec 2009 18:31:04 +0100 Subject: [PATCH 10/11] ALSA: fix incorrect rounding direction in snd_interval_ratnum() The direction of rounding is incorrect in the snd_interval_ratnum() It was detected with following parameters (sb8 driver playing 8kHz stereo file): - num is always 1000000 - requested frequency rate is from 7999 to 7999 (single frequency) The first loop calculates div_down(num, freq->min) which is 125. Thus, a frequency range's minimum value is 1000000 / 125 = 8000 Hz. The second loop calculates div_up(num, freq->max) which is 126 The frequency range's maximum value is 1000000 / 126 = 7936 Hz. The range maximum is lower than the range minimum so the function fails due to empty result range. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f410832a2..a27545b23ee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -758,7 +758,7 @@ int snd_interval_ratnum(struct snd_interval *i, int diff; if (q == 0) q = 1; - den = div_down(num, q); + den = div_up(num, q); if (den < rats[k].den_min) continue; if (den > rats[k].den_max) @@ -794,7 +794,7 @@ int snd_interval_ratnum(struct snd_interval *i, i->empty = 1; return -EINVAL; } - den = div_up(num, q); + den = div_down(num, q); if (den > rats[k].den_max) continue; if (den < rats[k].den_min) From db8cf334f66bdf1ba2b3d2f7128095fc9b7a6e2b Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 20:15:19 +0100 Subject: [PATCH 11/11] ALSA: sbawe: fix memory detection Memory amount is increased before a successful write-read sequence is done. Thus, 512 kB of onboard memory is detected on memoryless cards like SB32. Move the increasing of memory counter after successful read is done. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 96678d5d383..751762f1c59 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -393,8 +393,6 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { - size += 512 * 1024; /* increment 512kbytes */ - /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +412,9 @@ size_dram(struct snd_emu8000 *emu) /*snd_emu8000_read_wait(emu);*/ EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) - break; /* we must have wrapped around */ + break; /* no memory at this address */ + + size += 512 * 1024; /* increment 512kbytes */ snd_emu8000_read_wait(emu);