July 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] I2S simple driver
by Alessio Sangalli 31 Jul '08

31 Jul '08

Hi, we are writing a small driver to control a I2S interface on our ARM-based system... Right now we are going to implement a "easier" interface that allows to set some parameters, and then simply write to the device that will automatically control the DAC and then to the speakers. Now, this is of course not ALSA compliant. How can we estimate the effort to make this simple driver ALSA compatible so that every ALSA application can use this device for playback? We do not have any volume control and the range of sound "modes" is extremely limited. bye as

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[alsa-devel] Proper way to set SCMS copy prohibited bits on IEC958 S/PDIF
by Andrew Paprocki 31 Jul '08

31 Jul '08

I briefly checked the code for the Delta1010 (ice1712) to see if there was a way from user space to force the SCMS copy prohibited bits on in the S/PDIF stream. I see the driver initializes the bits to 0. Is there already a way to do this from user space, or would I have to modify the driver to force them on (or use a module parameter)? Thanks, -Andrew

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Re: [alsa-devel] [PATCH] pxa2xx-lib: support building for several pxa's
by Dmitry 31 Jul '08

31 Jul '08

Hi, 2008/7/30 Russ Dill <russ.dill(a)gmail.com>: > Here's a version of the third patch in the series without all the > #ifdef's. Just a quick question for Dmitry, why all the if (! > cpu_is_pxa3xx())'s around the pxa_gpio_mode's? Also, there was a build > error when just building the sound/arm pxa2xx-ac97. Good! AFAIR pxa_gpio_mode isn't valid for the the pxa3xx, so cpu_is_pxa3xx(). > > Support building pxa2xx-lib for several pxa chip versions by making code > run-time selected, not only compile-time > > Signed-off-by: Russ Dill <russ.dill(a)gmail.com> > --- > include/asm-arm/arch-pxa/pxa-regs.h | 4 +- > sound/arm/Kconfig | 2 +- > sound/arm/pxa2xx-ac97-lib.c | 174 ++++++++++++++++------------------ > sound/soc/pxa/pxa2xx-i2s.c | 16 ++- > 4 files changed, 95 insertions(+), 101 deletions(-) > -- With best wishes Dmitry

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Re: [alsa-devel] [PATCH] pxa2xx-lib: support building for several pxa's
by Eric Miao 31 Jul '08

31 Jul '08

> -----Original Message----- > From: Russ Dill [mailto:russ.dill@gmail.com] > Sent: Thursday, July 31, 2008 7:29 AM > To: Dmitry > Cc: alsa-devel(a)alsa-project.org; broonie(a)opensource.wolfsonmicro.com; Eric > Miao > Subject: Re: [PATCH] pxa2xx-lib: support building for several pxa's > > On Wed, Jul 30, 2008 at 3:41 PM, Dmitry <dbaryshkov(a)gmail.com> wrote: > > Hi, > > > > 2008/7/30 Russ Dill <russ.dill(a)gmail.com>: > >> Here's a version of the third patch in the series without all the > >> #ifdef's. Just a quick question for Dmitry, why all the if (! > >> cpu_is_pxa3xx())'s around the pxa_gpio_mode's? Also, there was a build > >> error when just building the sound/arm pxa2xx-ac97. > > > > Good! > > > > AFAIR pxa_gpio_mode isn't valid for the the pxa3xx, so cpu_is_pxa3xx(). > > I can't find any reason within the source that it wouldn't work. pxafb > uses it, which is used by littleton and zylonite. I'm not sure when > pxa_gpio_mode should be used vs the mfp pin config. > pxa_gpio_mode() is deprecated, please don't use that, even with pxa2xx platforms. The reason some legacy pxa_gpio_mode() is still there, e.g. pxafb, is that not all platforms have transitioned into use the new MFP API, til then, I'll schedule a cleanup to remove all those pxa_gpio_mode, so whenever possible, don't introduce any new pxa_gpio_mode(). P.S. the reason pxa_gpio_mode() doesn't cause any obvious consequence is that PXA3xx no longer has those GAFR registers, but that's really going to affect other registers, esp. the GPDR, which isn't correct. > >> > >> Support building pxa2xx-lib for several pxa chip versions by making code > >> run-time selected, not only compile-time > >> > >> Signed-off-by: Russ Dill <russ.dill(a)gmail.com> > >> --- > >> include/asm-arm/arch-pxa/pxa-regs.h | 4 +- > >> sound/arm/Kconfig | 2 +- > >> sound/arm/pxa2xx-ac97-lib.c | 174 > ++++++++++++++++------------------ > >> sound/soc/pxa/pxa2xx-i2s.c | 16 ++- > >> 4 files changed, 95 insertions(+), 101 deletions(-) > >> > > -- > > With best wishes > > Dmitry > >

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[alsa-devel] device_create_drvdata missing on older kernels
by Eliot Blennerhassett 31 Jul '08

31 Jul '08

I could not compile current git. (Error on acore/init.c) Adding this to adriver.h fixed it for me. I'm not sure if this is the correct general fix? #ifndef CONFIG_SND_HAS_DEVICE_CREATE_DRVDATA #define device_create_drvdata(c,d,m,p,s,n) device_create((c),(d),(m),(s),(n)) #endif (My kernel 2.6.24-19-generic) -- -- Eliot Blennerhassett www.audioscience.com

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Re: [alsa-devel] alsa-lib has anyone ever looked at taking it cross platform?
by Peter Dolding 30 Jul '08

30 Jul '08

> Peter Dolding wrote: >> This would kinda make logical sence. >> >> Alsa-lib plugin system in theory should be able to take any kind sound >> output under it. Simplify development of all applications on top of >> it. If alsa was cross platform why would you use something else as a >> middle body wrapper when you port. >> >> Windows and Mac support could be done as plugins. >> >> Or is there something that is in the alsa-lib that is a problem? >> >> Its all about reducing distance between application and hardware and >> reduce amount of coding. > > I'd rather see pulseaudio go fully cross platform. It's got some older > windows ports but if pulse were to get a push in the right direction for > windows and OSX then this lets alsa get on with the job of interfacing > with the audio on linux (that's what the L stands for after all), and > let higher level applications work with a more appropriate system with > more appropriate interfaces for practical use. Ok sorry to say the idea of let sound server do it is the same as saying we will put up for every with xlib being single threaded and have the toolkits above it deal with it. Instead of taking on the problem. xcb is taking on the problem in the X11 stack of libs. Except its many times worse. Pulseaudio and all other Sound Servers since its a extra service adds a extra point of complete sound failure to the Linux sound stack or what ever stack they happen to be operating on top of. It more of a hack around the limitations of the alsa framework. ALSA was really designed to avoid the need of such extras. The existence of Pulseaudio is really sign of failure. Reason Sound Servers existed at first due to the failures of OSS to provide a multi application sound support. ALSA was really designed to do away for the need of sound servers completely. Now alsa is failing to provide per application sound control and a sound server is back for another pass. Now why are not alsa interfaces practical to use. It is ment to be "Advanced Linux Sound Architecture" The complete Architecture. Not something you have to tack bits onto because its not functional or hard to use. Have we got to the point like xlib where the complete alsa-lib has to be reworked for usability. Now if you want to keep on thinking along that line has ALSA got to the point it should be dumped like OSS has been dumped because of its limitations? alsa-lib to freebsd or somewhere with oss most likely could be done really quickly since we already have a oss output option. Its also like saying the Linux Standard Base is only used by Linux. Its not its also used by a few non Linux's. Driver section of ALSA most likely will always be Linux only. But the wrapper layer that hides the hardware from the software really does not have to stay Linux only. Alsa is going up to be part of the Linux Standard Base its about time we all start looking at how it can be made most useful. Provide all the features it should and work for stability. Has anyone one attempted it or not. I guess not. Peter Dolding

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[alsa-devel] [PATCH] Permit simultaneous compilation of both PXA AC97 drivers
by Dmitry Baryshkov 30 Jul '08

30 Jul '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2df03ee..45ac468 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -360,7 +360,7 @@ int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_soc_platform pxa2xx_soc_platform = { .name = "pxa2xx-audio", .pcm_ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_pcm_new, + .pcm_new = pxa2xx_soc_pcm_new, .pcm_free = pxa2xx_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); -- 1.5.6.2

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[alsa-devel] [PATCH 1/1] ALSA: Fix limit of 8 PCM devices in SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE
by Pawel MOLL 30 Jul '08

30 Jul '08

ALSA: Fix limit of 8 PCM devices in SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine to have more than 8 PCM devices per card, except one place - the SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate devices > 7. This patch fixes the issue, changing the devices list organisation. Instead of adding new device to the tail, the list is now kept always ordered (by card number, then device number). Thus, during enumeration, it is easy to discover the fact that there is no more given card's devices. The same limit was present in OSS emulation code. It has been fixed as well. Additionally the device field of struct snd_pcm is now int, instead of unsigned int, as there is no obvious reason for keeping it unsigned. This caused a lot of problems with comparing this value with other (almost always signed) variables. There is just one more place where device number is unsigned - in struct snd_pcm_info, which should be also sorted out in future. Signed-off-by: Pawel MOLL <pawel.moll(a)st.com> diff --git a/include/sound/minors.h b/include/sound/minors.h index 46bcd20..a81798a 100644 --- a/include/sound/minors.h +++ b/include/sound/minors.h @@ -21,6 +21,8 @@ * */ +#define SNDRV_OS_MINORS 256 + #define SNDRV_MINOR_DEVICES 32 #define SNDRV_MINOR_CARD(minor) ((minor) >> 5) #define SNDRV_MINOR_DEVICE(minor) ((minor) & 0x001f) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 51d58cc..bfc096a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -84,8 +84,6 @@ struct snd_pcm_ops { * */ -#define SNDRV_PCM_DEVICES 8 - #define SNDRV_PCM_IOCTL1_FALSE ((void *)0) #define SNDRV_PCM_IOCTL1_TRUE ((void *)1) @@ -416,7 +414,7 @@ struct snd_pcm_str { struct snd_pcm { struct snd_card *card; struct list_head list; - unsigned int device; /* device number */ + int device; /* device number */ unsigned int info_flags; unsigned short dev_class; unsigned short dev_subclass; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 4c601b1..4ccd761 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2947,7 +2947,7 @@ static void register_oss_dsp(struct snd_pcm *pcm, int index) static int snd_pcm_oss_register_minor(struct snd_pcm *pcm) { pcm->oss.reg = 0; - if (dsp_map[pcm->card->number] == (int)pcm->device) { + if (dsp_map[pcm->card->number] == pcm->device) { char name[128]; int duplex; register_oss_dsp(pcm, 0); @@ -2963,7 +2963,7 @@ static int snd_pcm_oss_register_minor(struct snd_pcm *pcm) pcm->oss.reg++; pcm->oss.reg_mask |= 1; } - if (adsp_map[pcm->card->number] == (int)pcm->device) { + if (adsp_map[pcm->card->number] == pcm->device) { register_oss_dsp(pcm, 1); pcm->oss.reg++; pcm->oss.reg_mask |= 2; @@ -2988,7 +2988,7 @@ static int snd_pcm_oss_disconnect_minor(struct snd_pcm *pcm) snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_PCM, pcm->card, 1); } - if (dsp_map[pcm->card->number] == (int)pcm->device) { + if (dsp_map[pcm->card->number] == pcm->device) { #ifdef SNDRV_OSS_INFO_DEV_AUDIO snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_AUDIO, pcm->card->number); #endif @@ -3019,12 +3019,12 @@ static int __init alsa_pcm_oss_init(void) /* check device map table */ for (i = 0; i < SNDRV_CARDS; i++) { - if (dsp_map[i] < 0 || dsp_map[i] >= SNDRV_PCM_DEVICES) { + if (dsp_map[i] < 0 || dsp_map[i] >= SNDRV_OS_MINORS) { snd_printk(KERN_ERR "invalid dsp_map[%d] = %d\n", i, dsp_map[i]); dsp_map[i] = 0; } - if (adsp_map[i] < 0 || adsp_map[i] >= SNDRV_PCM_DEVICES) { + if (adsp_map[i] < 0 || adsp_map[i] >= SNDRV_OS_MINORS) { snd_printk(KERN_ERR "invalid adsp_map[%d] = %d\n", i, adsp_map[i]); adsp_map[i] = 1; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9dd9bc7..df4c8a3 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -42,7 +42,7 @@ static int snd_pcm_dev_free(struct snd_device *device); static int snd_pcm_dev_register(struct snd_device *device); static int snd_pcm_dev_disconnect(struct snd_device *device); -static struct snd_pcm *snd_pcm_search(struct snd_card *card, int device) +static inline struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) { struct snd_pcm *pcm; @@ -53,6 +53,37 @@ static struct snd_pcm *snd_pcm_search(struct snd_card *card, int device) return NULL; } +static inline int snd_pcm_next(struct snd_card *card, int device) +{ + struct snd_pcm *pcm; + + list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->card == card && pcm->device > device) + return pcm->device; + else if (pcm->card->number > card->number) + return -1; + } + return -1; +} + +static inline int snd_pcm_add(struct snd_pcm *newpcm) +{ + struct snd_pcm *pcm; + + list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->card == newpcm->card && pcm->device == newpcm->device) + return -EBUSY; + if (pcm->card->number > newpcm->card->number || + (pcm->card == newpcm->card && + pcm->device > newpcm->device)) { + list_add(&newpcm->list, pcm->list.prev); + return 0; + } + } + list_add_tail(&newpcm->list, &snd_pcm_devices); + return 0; +} + static int snd_pcm_control_ioctl(struct snd_card *card, struct snd_ctl_file *control, unsigned int cmd, unsigned long arg) @@ -65,14 +96,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; - while (device < SNDRV_PCM_DEVICES) { - if (snd_pcm_search(card, device)) - break; - device++; - } - if (device == SNDRV_PCM_DEVICES) - device = -1; + device = snd_pcm_next(card, device); mutex_unlock(&register_mutex); if (put_user(device, (int __user *)arg)) return -EFAULT; @@ -98,7 +122,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(&register_mutex); - pcm = snd_pcm_search(card, device); + pcm = snd_pcm_get(card, device); if (pcm == NULL) { err = -ENXIO; goto _error; @@ -931,11 +955,11 @@ static int snd_pcm_dev_register(struct snd_device *device) snd_assert(pcm != NULL && device != NULL, return -ENXIO); mutex_lock(&register_mutex); - if (snd_pcm_search(pcm->card, pcm->device)) { + err = snd_pcm_add(pcm); + if (err) { mutex_unlock(&register_mutex); - return -EBUSY; + return err; } - list_add_tail(&pcm->list, &snd_pcm_devices); for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; if (pcm->streams[cidx].substream == NULL) diff --git a/sound/core/sound.c b/sound/core/sound.c index 1003ae3..838dd9e 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -34,8 +34,6 @@ #include <linux/kmod.h> #include <linux/mutex.h> -#define SNDRV_OS_MINORS 256 - static int major = CONFIG_SND_MAJOR; int snd_major; EXPORT_SYMBOL(snd_major); -- 1.5.5.1

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[alsa-devel] multiple identical controls
by Jon Smirl 30 Jul '08

30 Jul '08

My hardware has four channels, each with seven identical biquads. I'm making a custom control to handle them. Don't I want to give then an index from 0 to 6? Why is snd_soc_cnew() setting index to zero? struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, char *long_name) { struct snd_kcontrol_new template; memcpy(&template, _template, sizeof(template)); if (long_name) template.name = long_name; template.index = 0; return snd_ctl_new1(&template, data); } EXPORT_SYMBOL_GPL(snd_soc_cnew); -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH v2] asoc: restrict sample rate and size in Freescale MPC8610 sound drivers
by Timur Tabi 29 Jul '08

29 Jul '08

The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Please apply this patch for 2.6.26.1 and for 2.6.27-rc1. Without this patch, recording on the MPC8610 is just plain broken, so it is a must-fix for 2.6.26. sound/soc/fsl/fsl_dma.c | 7 ++++- sound/soc/fsl/fsl_ssi.c | 74 ++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 70 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc59..7ceea2b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33..157a789 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,8 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: pointer to the stream that was opened first + * @second_stream: pointer to second stream * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +81,8 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) */ } + if (!ssi_private->first_stream) + ssi_private->first_stream = substream; + else { + /* This is the second stream open, so we need to impose sample + * rate and maybe sample size constraints. Note that this can + * cause a race condition if the second stream is opened before + * the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); + + ssi_private->second_stream = substream; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ssi_private->playback++; @@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ssi_private->capture--; + if (ssi_private->first_stream == substream) + ssi_private->first_stream = ssi_private->second_stream; + + ssi_private->second_stream = NULL; + /* * If this is the last active substream, disable the SSI and release * the IRQ. -- 1.5.5

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