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Re: [alsa-devel] [PATCH 1/1] ASoC: Samsung: Convert to use snd_soc_register_card()
by Sangbeom Kim 03 Jul '12

03 Jul '12

Hi! On Tuesday, Jul 03, 2012 at 05:34PM, Sachin Kamat wrote: > Current method for machine driver to register with the ASoC core is to use > snd_soc_register_card() instead of creating a "soc-audio" platform device. It looks better to include reason of using snd_soc_register_card instead of creating "soc-audio" or warning message. Acked-by: Sangbeom Kim <sbkim73(a)samsung.com>

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Mark Brown 03 Jul '12

03 Jul '12

On Tue, Jul 03, 2012 at 11:59:31AM +0530, Laxman Dewangan wrote: > Are you taking this patch? > I have other independent patch for the writecombine dma support in > pcm memory library and wanted to avoid any merge conflicts. > So if it goes in your tree then the other patch also need to go in > your tree. You've not even left 24 hours for me to respond!

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Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 03 Jul '12

03 Jul '12

At Tue, 3 Jul 2012 10:20:13 +0200, Philipp Dreimann wrote: > > On 3 July 2012 10:00, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 3 Jul 2012 09:41:09 +0200, > > Philipp Dreimann wrote: > >> > >> On 2 July 2012 14:10, Daniel Mack <zonque(a)gmail.com> wrote: > >> > On 01.07.2012 13:40, Philipp Dreimann wrote: > >> >> Hello, > >> >> > >> >> my usb webcam, which happens to have a microphone too, stopped working > >> >> using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) > >> > > >> > Please try 3.5-rc5. A number of regressions were fixed just recently. > >> > > >> > Let me know if that version still causes problems. > >> > >> It still does. > > > > Try to pass ignore_ctl_error=1 to snd-usb-audio module. > > If this works and no other problem happens, we can add a similar quirk > > like some Logitech webcams. > > This does not seem to help. And webcam itself is working with our without snd-usb-audio module? Which device at all? Too little information to analyze. Takashi

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Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 03 Jul '12

03 Jul '12

At Tue, 3 Jul 2012 09:41:09 +0200, Philipp Dreimann wrote: > > On 2 July 2012 14:10, Daniel Mack <zonque(a)gmail.com> wrote: > > On 01.07.2012 13:40, Philipp Dreimann wrote: > >> Hello, > >> > >> my usb webcam, which happens to have a microphone too, stopped working > >> using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) > > > > Please try 3.5-rc5. A number of regressions were fixed just recently. > > > > Let me know if that version still causes problems. > > It still does. Try to pass ignore_ctl_error=1 to snd-usb-audio module. If this works and no other problem happens, we can add a similar quirk like some Logitech webcams. Takashi

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[alsa-devel] [PATCH] ASoC: wm8962: Log AIF configuration requested by hw_params()
by Mark Brown 02 Jul '12

02 Jul '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index aa5839c..5781581 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2582,6 +2582,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); + dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n", + wm8962->bclk, wm8962->lrclk); + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) wm8962_configure_bclk(codec); -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 02 Jul '12

02 Jul '12

On 07/02/2012 02:25 AM, Laxman Dewangan wrote: > On Saturday 30 June 2012 08:38 PM, Laxman Dewangan wrote: >> On Friday 29 June 2012 10:32 PM, Stephen Warren wrote: >>> On 06/29/2012 05:34 AM, Laxman Dewangan wrote: >>>> Use the dmaengine based Tegra APB DMA driver for >>>> data transfer between SPI fifo and memory in >>>> place of legacy Tegra APB DMA. >>>> >>>> Because generic soc-dmaengine-pcm uses the DMAs API >>>> based on dmaengine, using the exported APIs provided >>>> by this generic driver. >>>> >>>> The new driver is selected if legacy driver is not >>>> selected and new dma driver is enabled through config >>>> file. >>> >>> This works just fine with the existing non-dmaengine DMA driver enabled. >>> >>> However, I can't get it to work with dmaengine: ... >> I think I should send that patches so that you can test it by: >> - Taking already applied dma driver change in linux-next. >> - Apply my new patches which I am going to send. >> >> And do local change in the tegra_defconfig to disable SYSTEM_DMA and >> enable dmaengine based dma driver. > > The new dma patches are: > > dma: tegra: fix residual calculation for cyclic case > dma: tegra: rename driver and compatible to match with dts > ... OK, with those two patches plus this one, everything works on Cardhu/Tegra30 using dmaengine. However, on Tegra20 DMA doesn't appear to work. No audio is heard, and aplay eventually exits with the following error message: aplay: pcm_write:1603: write error: Input/output error I believe this means that the DMA simply isn't operating. I'll assume this is a Tegra20-specific issue in the dmaengine driver itself, and hence not a problem with this patch, so I'll ack this patch.

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 02 Jul '12

02 Jul '12

On 06/29/2012 05:34 AM, Laxman Dewangan wrote: > Use the dmaengine based Tegra APB DMA driver for > data transfer between SPI fifo and memory in > place of legacy Tegra APB DMA. > > Because generic soc-dmaengine-pcm uses the DMAs API > based on dmaengine, using the exported APIs provided > by this generic driver. > > The new driver is selected if legacy driver is not > selected and new dma driver is enabled through config > file. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Acked-by: Stephen Warren <swarren(a)wwwdotorg.org> Tested-by: Stephen Warren <swarren(a)wwwdotorg.org> (Tested on Cardhu/Tegra30. It fails on Whistler/Tegra20, but I assume that's due to a bug in the dmaengine driver not this patch. There won't be a regression when applying this patch since we haven't switched to dmaengine as the default yet).

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Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 02 Jul '12

02 Jul '12

On 01.07.2012 13:40, Philipp Dreimann wrote: > Hello, > > my usb webcam, which happens to have a microphone too, stopped working > using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) Please try 3.5-rc5. A number of regressions were fixed just recently. Let me know if that version still causes problems. Daniel

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[alsa-devel] 24 bit sample embedded in 32 bits
by Vipin Kumar 02 Jul '12

02 Jul '12

Hello, I assume that the alsa framework expects the 24 bit sample data to be packed one after the other. for example while recording a 2 channel, 24 bit sample data, typically the dma reads a fifo (which receives data from adc) and prepares a buffer which is then passed to the upper layers. This buffer is expeted to be 24byte left channel and then 24 byte right channel. Am I right? If the above assumption is correct, I need to find a way to tell the framework that the data is 24bit but is embedded in 32 bit. ie 32bit left and then 32 bit right channel data I am tending to think that there has to be a way but I am unable to find it. Can you please point me to the right direction Regards Vipin

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Re: [alsa-devel] [PATCH 2/2] ASoC: tegra: use sound pcm library for substream dma buffer
by Takashi Iwai 02 Jul '12

02 Jul '12

At Mon, 2 Jul 2012 14:24:33 +0530, Laxman Dewangan wrote: > > The sound pcm library support the writecombine DMA buffer > allocation/deallocation/mapping and hence using the APIs > form sound pcm library inplace of implementing the same > locally. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > Changes from V1: > Changes done to use the existing APIs for sound pcm memory management > which is now supporting writecombine. > > sound/soc/tegra/tegra_pcm.c | 112 ++++++++++-------------------------------- > 1 files changed, 27 insertions(+), 85 deletions(-) > > diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c > index 127348d..ec1fa25 100644 > --- a/sound/soc/tegra/tegra_pcm.c > +++ b/sound/soc/tegra/tegra_pcm.c > @@ -202,8 +202,14 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, > { > struct snd_pcm_runtime *runtime = substream->runtime; > struct tegra_runtime_data *prtd = runtime->private_data; > + int ret; > > - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); > + ret = snd_pcm_lib_malloc_pages(substream, > + params_buffer_bytes(params)); > + if (ret < 0) { > + pr_err("page allocation failed, err %d\n", ret); Do you really want to spew error messages at each time? > + return ret; > + } > > prtd->dma_req[0].size = params_period_bytes(params); > prtd->dma_req[1].size = prtd->dma_req[0].size; > @@ -213,9 +219,7 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, > > static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) > { > - snd_pcm_set_runtime_buffer(substream, NULL); > - > - return 0; > + return snd_pcm_lib_free_pages(substream); > } > > static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > @@ -263,18 +267,6 @@ static snd_pcm_uframes_t tegra_pcm_pointer(struct snd_pcm_substream *substream) > return prtd->period_index * runtime->period_size; > } > > - > -static int tegra_pcm_mmap(struct snd_pcm_substream *substream, > - struct vm_area_struct *vma) > -{ > - struct snd_pcm_runtime *runtime = substream->runtime; > - > - return dma_mmap_writecombine(substream->pcm->card->dev, vma, > - runtime->dma_area, > - runtime->dma_addr, > - runtime->dma_bytes); > -} > - > static struct snd_pcm_ops tegra_pcm_ops = { > .open = tegra_pcm_open, > .close = tegra_pcm_close, > @@ -283,87 +275,37 @@ static struct snd_pcm_ops tegra_pcm_ops = { > .hw_free = tegra_pcm_hw_free, > .trigger = tegra_pcm_trigger, > .pointer = tegra_pcm_pointer, > - .mmap = tegra_pcm_mmap, > + .mmap = snd_pcm_lib_default_mmap, You can simply remove .mmap definition. NULL falls back to the default mmap handler. > }; > > -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) > -{ > - struct snd_pcm_substream *substream = pcm->streams[stream].substream; > - struct snd_dma_buffer *buf = &substream->dma_buffer; > - size_t size = tegra_pcm_hardware.buffer_bytes_max; > - > - buf->area = dma_alloc_writecombine(pcm->card->dev, size, > - &buf->addr, GFP_KERNEL); > - if (!buf->area) > - return -ENOMEM; > - > - buf->dev.type = SNDRV_DMA_TYPE_DEV; > - buf->dev.dev = pcm->card->dev; > - buf->private_data = NULL; > - buf->bytes = size; > - > - return 0; > -} > - > -static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) > -{ > - struct snd_pcm_substream *substream; > - struct snd_dma_buffer *buf; > - > - substream = pcm->streams[stream].substream; > - if (!substream) > - return; > - > - buf = &substream->dma_buffer; > - if (!buf->area) > - return; > - > - dma_free_writecombine(pcm->card->dev, buf->bytes, > - buf->area, buf->addr); > - buf->area = NULL; > -} > - > -static u64 tegra_dma_mask = DMA_BIT_MASK(32); > - > +#define TEGRA_PCM_PREALLOC_BUFFER (32 * 1024) > +#define TEGRA_PCM_PREALLOC_BUFFER_MAX (32 * 1024) > static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > - struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > - > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &tegra_dma_mask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); The mask setup is still required even if you use the preallocation helper. > - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > - ret = tegra_pcm_preallocate_dma_buffer(pcm, > - SNDRV_PCM_STREAM_PLAYBACK); > - if (ret) > - goto err; > - } > - > - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { > - ret = tegra_pcm_preallocate_dma_buffer(pcm, > - SNDRV_PCM_STREAM_CAPTURE); > - if (ret) > - goto err_free_play; > + struct snd_card *card = rtd->card->snd_card; > + int retval = 0; > + > + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || > + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { You can omit this check. snd_pcm_lib_preallocate_pages_for_all() will ignore the streams that have no substreams assigned. Takashi > + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, > + SNDRV_DMA_TYPE_DEV_WC, > + card->dev, > + TEGRA_PCM_PREALLOC_BUFFER, > + TEGRA_PCM_PREALLOC_BUFFER_MAX); > + if (retval < 0) > + dev_err(card->dev, > + "dma buffer pre-alloc failied %d\n", retval); > } > - > - return 0; > - > -err_free_play: > - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); > -err: > - return ret; > + return retval; > } > > static void tegra_pcm_free(struct snd_pcm *pcm) > { > - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); > - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); > + snd_pcm_lib_preallocate_free_for_all(pcm); > } > > + > static struct snd_soc_platform_driver tegra_pcm_platform = { > .ops = &tegra_pcm_ops, > .pcm_new = tegra_pcm_new, > -- > 1.7.1.1 >

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Re: [alsa-devel] [PATCH V2 1/2] ALSA: Support writecombine DMA buffer memory page alloc/free
by Takashi Iwai 02 Jul '12

02 Jul '12

At Mon, 2 Jul 2012 14:24:32 +0530, Laxman Dewangan wrote: > > Some of architecture like ARM support the writecombine DMA > buffer which is used for pcm substream DMA buffer. > > Supporting writecombine DMA buffer allocation/free/mmap > through the sound library memory management. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > Changes from V1: > Integrate the changes from Takashi's sample code for supporting > writecombine. Could you give a short comment in each file that ifdef CONFIG_ARM is a workaround? thanks, Takashi > > include/sound/memalloc.h | 1 + > sound/core/memalloc.c | 74 +++++++++++++++++++++++++++++++++++++++++----- > sound/core/pcm_native.c | 10 ++++++ > 3 files changed, 77 insertions(+), 8 deletions(-) > > diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h > index c425062..0328e37 100644 > --- a/include/sound/memalloc.h > +++ b/include/sound/memalloc.h > @@ -52,6 +52,7 @@ struct snd_dma_device { > #else > #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ > #endif > +#define SNDRV_DMA_TYPE_DEV_WC 4 /* writecombine page */ > > /* > * info for buffer allocation > diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c > index 6915692..7468251 100644 > --- a/sound/core/memalloc.c > +++ b/sound/core/memalloc.c > @@ -124,8 +124,10 @@ void snd_free_pages(void *ptr, size_t size) > */ > > #ifdef CONFIG_HAS_DMA > -/* allocate the coherent DMA pages */ > -static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) > +static void *_snd_malloc_dev_pages(struct device *dev, size_t size, > + dma_addr_t *dma, > + void *(*alloc_func)(struct device *dev, size_t size, > + dma_addr_t *dma, gfp_t gfp_flags)) > { > int pg; > void *res; > @@ -138,16 +140,18 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d > | __GFP_COMP /* compound page lets parts be mapped */ > | __GFP_NORETRY /* don't trigger OOM-killer */ > | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ > - res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); > + res = alloc_func(dev, PAGE_SIZE << pg, dma, gfp_flags); > if (res != NULL) > inc_snd_pages(pg); > > return res; > } > > -/* free the coherent DMA pages */ > -static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > - dma_addr_t dma) > +static void _snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > + dma_addr_t dma, > + void (*free_func)(struct device *dev, > + size_t size, void *vaddr, > + dma_addr_t dma_handle)) > { > int pg; > > @@ -155,8 +159,51 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > return; > pg = get_order(size); > dec_snd_pages(pg); > - dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); > + free_func(dev, PAGE_SIZE << pg, ptr, dma); > +} > + > +static void *_dma_alloc_coherent(struct device *dev, size_t size, > + dma_addr_t *dma_handle, gfp_t flag) > +{ > + return dma_alloc_coherent(dev, size, dma_handle, flag); > +} > + > +static void _dma_free_coherent(struct device *dev, size_t size, > + void *vaddr, dma_addr_t dma_handle) > +{ > + dma_free_coherent(dev, size, vaddr, dma_handle); > +} > + > + > +/* allocate the coherent DMA pages */ > +static void *snd_malloc_dev_pages(struct device *dev, size_t size, > + dma_addr_t *dma) > +{ > + return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > +} > + > +/* free the coherent DMA pages */ > +static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > + dma_addr_t dma) > +{ > + _snd_free_dev_pages(dev, size, ptr, dma, _dma_free_coherent); > } > + > +#ifdef CONFIG_ARM > +/* allocate the writecombine DMA pages */ > +static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, > + dma_addr_t *dma) > +{ > + return _snd_malloc_dev_pages(dev, size, dma, dma_alloc_writecombine); > +} > + > +/* free the writecombine DMA pages */ > +static void snd_free_dev_wc_pages(struct device *dev, size_t size, void *ptr, > + dma_addr_t dma) > +{ > + _snd_free_dev_pages(dev, size, ptr, dma, dma_free_writecombine); > +} > +#endif > #endif /* CONFIG_HAS_DMA */ > > /* > @@ -200,6 +247,11 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, > case SNDRV_DMA_TYPE_DEV: > dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); > break; > +#ifdef CONFIG_ARM > + case SNDRV_DMA_TYPE_DEV_WC: > + dmab->area = snd_malloc_dev_wc_pages(device, size, &dmab->addr); > + break; > +#endif > #endif > #ifdef CONFIG_SND_DMA_SGBUF > case SNDRV_DMA_TYPE_DEV_SG: > @@ -272,6 +324,12 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) > case SNDRV_DMA_TYPE_DEV: > snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); > break; > +#ifdef CONFIG_ARM > + case SNDRV_DMA_TYPE_DEV_WC: > + snd_free_dev_wc_pages(dmab->dev.dev, dmab->bytes, dmab->area, > + dmab->addr); > + break; > +#endif > #endif > #ifdef CONFIG_SND_DMA_SGBUF > case SNDRV_DMA_TYPE_DEV_SG: > @@ -378,7 +436,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) > long pages = snd_allocated_pages >> (PAGE_SHIFT-12); > struct snd_mem_list *mem; > int devno; > - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; > + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" , "WC"}; > > mutex_lock(&list_mutex); > seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 53b5ada..28d949c 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -3170,6 +3170,16 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, > struct vm_area_struct *area) > { > area->vm_flags |= VM_RESERVED; > + > +#ifdef CONFIG_ARM > + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_WC) > + return dma_mmap_writecombine(substream->dma_buffer.dev.dev, > + area, > + substream->runtime->dma_area, > + substream->runtime->dma_addr, > + area->vm_end - area->vm_start); > +#endif > + > #ifdef ARCH_HAS_DMA_MMAP_COHERENT > if (!substream->ops->page && > substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) > -- > 1.7.1.1 >

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[alsa-devel] 2.6.18 unknown symbol ksize error
by Patrick Shirkey 02 Jul '12

02 Jul '12

Hi, I have compiled the latest stable alsa drivers against 2.6.18 kernel with gcc-4.1. I am getting an error when I insert the sound card modules. snd_rawmidi: Unknown symbol ksize As far as I can tell 2.6.18 has support for ksize so I am not sure where this error is coming from. -- Patrick Shirkey Boost Hardware Ltd

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[alsa-devel] alsa-driver cross compile problem - upgrading ALSA
by Max Klein 02 Jul '12

02 Jul '12

I am running 2.6.27 kernel on arm board with ALSA driver version 1.0.17 (cat /proc/asound/version) and I would like to update ALSA driver version to current 1.0.25. I am trying to build alsa-drivers with "./configure --with-cross=arm-none-linux-gnueabi- --enable-dynamic-minors --with-debug=full --with-isapnp=no --with-oss=no --with-kernel=/data/src/kernel/linux-davinci --with-build=/data/src/kernel/linux-davinci --with-sequencer=no" Afterwards I compile it via "make ARCH=arm" but this fails we the following error: In file included from /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/acore/hrtimer.c:1: /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/include/adriver.h:753: error: static declaration of 'jiffies_to_msecs' follows non-static declaration include/linux/jiffies.h:286: error: previous declaration of 'jiffies_to_msecs' was here /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/include/adriver.h:772: error: static declaration of 'msecs_to_jiffies' follows non-static declaration include/linux/jiffies.h:288: error: previous declaration of 'msecs_to_jiffies' was here Can someone please help me how to get ALSA upgraded on my platform? Obviously I am either doing something wrong when configuring or I am on the absolutely wrong way? I am looking forward for any advice, thanks in advance! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[alsa-devel] [PATCH] mioa701 convert to snd_soc_register_card API
by Robert Jarzmik 01 Jul '12

01 Jul '12

The mioa701 board code is converted to the snd_soc_register_card() and snd_soc_unregister_card() APIs. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/pxa/mioa701_wm9713.c | 33 ++++++++++++--------------------- 1 file changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 9c585af..8687c1c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = { .num_links = ARRAY_SIZE(mioa701_dai), }; -static struct platform_device *mioa701_snd_device; - -static int mioa701_wm9713_probe(struct platform_device *pdev) +static int __devinit mioa701_wm9713_probe(struct platform_device *pdev) { - int ret; + int rc; if (!machine_is_mioa701()) return -ENODEV; - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" - "lead to overheating and possible destruction of your device." - "Do not use without a good knowledge of mio's board design!\n"); - - mioa701_snd_device = platform_device_alloc("soc-audio", -1); - if (!mioa701_snd_device) - return -ENOMEM; - - platform_set_drvdata(mioa701_snd_device, &mioa701); - - ret = platform_device_add(mioa701_snd_device); - if (!ret) - return 0; - - platform_device_put(mioa701_snd_device); - return ret; + mioa701.dev = &pdev->dev; + rc = snd_soc_register_card(&mioa701); + if (!rc) + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + " Do not use without a good knowledge of mio's board design!\n"); + return rc; } static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) { - platform_device_unregister(mioa701_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; } -- 1.7.10

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[alsa-devel] [PATCH] ASoC: soc-dapm: Use ASYNC_DOMAIN_EXCLUSIVE
by Fabio Estevam 01 Jul '12

01 Jul '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit e895407d (async: introduce 'async_domain' type) changed the parameter types of async_schedule_domain() and async_synchronize_full_domain(). Use ASYNC_DOMAIN_EXCLUSIVE and fix the following build warnings: sound/soc/soc-dapm.c:1646:6: warning: passing argument 3 of 'async_schedule_domain' from incompatible pointer type sound/soc/soc-dapm.c:1647:2: warning: passing argument 1 of 'async_synchronize_full_domain' from incompatible pointer type sound/soc/soc-dapm.c:1660:6: warning: passing argument 3 of 'async_schedule_domain' from incompatible pointer type sound/soc/soc-dapm.c:1661:2: warning: passing argument 1 of 'async_synchronize_full_domain' from incompatible pointer type Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Build tested only. sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9670668..c93e832 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1572,7 +1572,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) struct snd_soc_dapm_context *d; LIST_HEAD(up_list); LIST_HEAD(down_list); - LIST_HEAD(async_domain); + ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; trace_snd_soc_dapm_start(card); -- 1.7.1

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Re: [alsa-devel] [pulseaudio-discuss] asoundrc, configure one virtual device for both input and output
by Tanu Kaskinen 01 Jul '12

01 Jul '12

(CC'ing alsa-devel, because I had problems with building alsa-plugins) On Sun, 2012-07-01 at 14:16 +0900, Amar Akshat wrote: > Help me here with two things, > > 1) How do I fix this and test, as in should I clone the alsa-project, > or download alsa code and fix and compile ? In case I have to clone > it, can you also provide me the clone instructions. git clone git://git.alsa-project.org/alsa-plugins.git alsa-plugins cd alsa-plugins ./gitcompile --enable-pulseaudio If the configure script complains about missing dependencies, you'll need to install them. After you have made your changes, I believe you should be able to test them by running "make && sudo make install". I haven't tried that myself. I tried the "./gitcompile --enable-pulseaudio" command myself, and had this problem: Making all in pulse make[2]: Entering directory `/home/tanu/dev/alsa-plugins/pulse' /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT ctl_pulse.lo -MD -MP -MF .deps/ctl_pulse.Tpo -c -o ctl_pulse.lo ctl_pulse.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT ctl_pulse.lo -MD -MP -MF .deps/ctl_pulse.Tpo -c ctl_pulse.c -fPIC -DPIC -o .libs/ctl_pulse.o mv -f .deps/ctl_pulse.Tpo .deps/ctl_pulse.Plo /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT pulse.lo -MD -MP -MF .deps/pulse.Tpo -c -o pulse.lo pulse.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT pulse.lo -MD -MP -MF .deps/pulse.Tpo -c pulse.c -fPIC -DPIC -o .libs/pulse.o mv -f .deps/pulse.Tpo .deps/pulse.Plo /bin/bash ../libtool --tag=CC --mode=link gcc -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -module -avoid-version -export-dynamic -no-undefined -o libasound_module_ctl_pulse.la -rpath /usr/lib/alsa-lib ctl_pulse.lo pulse.lo -lasound -L/usr/local/lib -lpulse -lasound libtool: link: gcc -shared -fPIC -DPIC .libs/ctl_pulse.o .libs/pulse.o -L/usr/local/lib /usr/local/lib/libpulse.so -lasound -O2 -pthread -Wl,-soname -Wl,libasound_module_ctl_pulse.so -o .libs/libasound_module_ctl_pulse.so /bin/sed: can't read /usr/lib/x86_64-linux-gnu/libsndfile.la: No such file or directory libtool: link: `/usr/lib/x86_64-linux-gnu/libsndfile.la' is not a valid libtool archive make[2]: *** [libasound_module_ctl_pulse.la] Error 1 make[2]: Leaving directory `/home/tanu/dev/alsa-plugins/pulse' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/tanu/dev/alsa-plugins' make: *** [all] Error 2 Debian doesn't ship libsndfile.la, so I tried installing libsndfile from source. That didn't help at all. I eventually figured that since the alsa build system doesn't reference libsndfile at all, libpulse must somehow have a reference to /usr/lib/x86_64-linux-gnu/libsndfile.la (even though my system has never had that file installed). I rebuilt pulseaudio and installed it, and now it works (apparently because the rebuilt version references the version of libsndfile that was built from source, which includes libsndfile.la, instead of referencing the system version of libsndfile). Does anyone know if the build system of alsa-plugins could be improved so that it doesn't use any external .la files when linking? > 2) and after my modifications, my config in asoundrc should look like this ? > > pcm.pulse_dev1 { > type pulse > source alsa_input.pci-0000_00_1b.0.analog-stereo > sink alsa_output.pci-0000_00_1b.0.analog-mono > } Yes. -- Tanu

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 29 Jun '12

29 Jun '12

On 06/29/2012 05:34 AM, Laxman Dewangan wrote: > Use the dmaengine based Tegra APB DMA driver for > data transfer between SPI fifo and memory in > place of legacy Tegra APB DMA. > > Because generic soc-dmaengine-pcm uses the DMAs API > based on dmaengine, using the exported APIs provided > by this generic driver. > > The new driver is selected if legacy driver is not > selected and new dma driver is enabled through config > file. This works just fine with the existing non-dmaengine DMA driver enabled. However, I can't get it to work with dmaengine: > # aplay ~/abba-dq-48000-stereo.wav > [ 151.613476] tegra20-i2s tegra20-i2s.0: dmaengine pcm open failed with err -6 > [ 151.620557] tegra20-i2s tegra20-i2s.0: can't open platform tegra20-i2s.0: -6 > aplay: main:654: audio open error: No such device or address I do have the following in my local tree: 68a67b8 ARM: tegra: add device tree AUXDATA for APBDMA 0db7a96 ARM: tegra: dma: rename driver name for clock to "tegra-apbdma" I also fixed the compatible values in drivers/dma/tegra20-apb-dma.c so the driver would get instantiated, which it does; /sys/devices/tegra-apbdma/dma has a bunch of dmaengine channels in it. (Note: This is on Ventana, although I doubt that makes much difference) Is there something else I need to do to test this?

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 29 Jun '12

29 Jun '12

At Thu, 28 Jun 2012 19:06:07 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 06:58 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 18:34:02 +0530, > > Laxman Dewangan wrote: > >> On Thursday 28 June 2012 06:27 PM, Takashi Iwai wrote: > >>> At Thu, 28 Jun 2012 18:05:53 +0530, > >>> Laxman Dewangan wrote: > >>>> On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > >>>>> * PGP Signed by an unknown key > >>>>> > >>>>> On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >>>>>> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>>>>>> Oh, dear - this means we don't actually have a generic API at all. Is > >>>>>>> there any effort being made to make this generally available? > >>>>>> Yes there is, and in fact in next/master there is generic > >>>>>> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >>>>>> though. But in theory it should fallback to a normal coherent mapping. Marek > >>>>>> did a lot of work on this, I've put him on Cc. > >>>> So can we put the function snd_pcm_lib_writecombine_mmap() in the > >>>> pcm_native.c and only export this api for ARM i.e. under macro #ifdef > >>>> CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > >>>> directly use that one? > >>>> Or, wait for this common API until all ARCH support it? > >>> I think it's fine to put it first in ALSA side with some ifdef. > >>> A similar trick is already found for snd_pcm_lib_mmap_iomem. > >>> See include/sound/pcm.h. > >>> > >>> But, actually it's a still question what if an architecture doesn't > >>> support the mmap of writecombine at all. The proposed patch doesn't > >>> allow you to know whether writecombine-mmap is possible or not on the > >>> running architecture until you really try to call it and fail. > >>> It's a missing piece, IMO. > >> Lars wanted to move the buffer allocation also to common place. > > Which common place? > > He wanted to refactor allocation part also at same time and so I guessed > that it will be in some common place and though about the sound/core. But the memory allocation rework is basically irrelevant from what's needed for pcm_mmap_writecombine() now. It'd need a lot more reworks than the simple addition of dma_mmap_writecombine() wrapper. > >> Then how about this? > >> create new file and header for snd-pcm-writecombine-buffer.c/.h and put > >> this in sound/core. > >> Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > >> which will be select this config through sound/soc/xxx/Kconfig if they > >> want to use. > >> This will provide three apis: new, free and mmap. > > Way too much hustles than necessary... > Ooh no. :-( I had taken the idea from snd_dmaengine_pcm driver. > Bit I like to hear simple way which solves the purpose. As Lars suggested, a simple ifdef should suffice for now. With the upcoming generic dma_mmap_writecombine() stuff, it'd be even simpler in future. But, still we need to be careful about this. As mentioned, there is no flag to know the possibility of writecombine mmap beforehand. It'd be nice if we have either a compile-time or a run-time flag / function to check that. Then the driver can also expose the mmap capability to user-space depending on the flag. Takashi

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Lars-Peter Clausen 29 Jun '12

29 Jun '12

On 06/29/2012 05:04 PM, Laxman Dewangan wrote: > [...] > > >> +/* allocate the coherent DMA pages */ >> +static void *snd_malloc_dev_pages(struct device *dev, size_t size, >> dma_addr_t *dma) >> +{ >> + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); > > This does not get compiled in ARM because dma_alloc_coherant is macro > defined as > > #define dma_alloc_coherent(d, s, h, f) dma_alloc_attrs(d, s, h, f, NULL) > > static inline void *dma_alloc_attrs(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag, > struct dma_attrs *attrs) > > > I fixed this by > static void *_dma_alloc_coherent(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag) > { > return dma_alloc_coherent(dev, size, dma_handle, flag); > } > > /* allocate the coherent DMA pages */ > static void *snd_malloc_dev_pages(struct device *dev, size_t size, > dma_addr_t *dma) > { > return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > } I think all dma_alloc_* functions use dma_alloc_attrs internally these days. I think it might be a better idea just to use dma_alloc_attrs in __snd_malloc_dev_pages and just pass a dma_attrs struct instead of a function. Something like: static void *__snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma, struct dma_attrs *attrs) { .... res = dma_alloc_attrs(dev, PAGE_SIZE << pg, dma, gfp_flags, attrs); .... } static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) { return __snd_malloc_dev_pages(dev, size, dma, NULL); } static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, dma_addr_t *dma) { DEFINE_DMA_ATTRS(attrs); dma_set_attr(DMA_ATTR_WRITE_COMBINE, &attrs); return __snd_malloc_dev_pages(dev, size, dma, &attrs); } Same applies for freeing the memory again. - Lars

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Takashi Iwai 29 Jun '12

29 Jun '12

At Fri, 29 Jun 2012 20:34:24 +0530, Laxman Dewangan wrote: > > Hi Takashi, > Thanks for sample code. It helps lot. > > On Friday 29 June 2012 05:43 PM, Takashi Iwai wrote: > > At Fri, 29 Jun 2012 15:53:15 +0530, > > Laxman Dewangan wrote: > >> Some of the ARM based soc allocate the writecombine dma buffer for > >> pcm substreams. They have the same codes for managing this buffer. > >> Moving this to the core/pcm files so that they can use that directly. > >> > >> Remove the code from Tegra PCM and use these new library function. > >> > >> This is enabled only for ARM specific and can be extended to other > >> architecture if they support the writecombine dma buffer. > >> > >> This patch is based on detail discussion on patch: > >> [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap > >> And suggestion from Lars and Takashi. > > Looking through your patch, I think an easier integration is just to > > add writecombine option to memalloc.c which calls > > dma_alloc_writecombine() instead of dma_alloc_coherent(). > > > > The addition for mmap is still an open question. Again, an easier > > option so far looks like just add the call of dma_alloc_writecombine() > > in snd_pcm_lib_default_mmap(). Alternatively, we can create an > > individual mmap pcm_ops as we discussed. > > > > Below is a totally untested patch, but you can imagine what I meant. > > > > After this change, replace with snd_pcm_lib_malloc_pages with > > SNDRV_DMA_TYPE_DEV_WC, and that's all you need in the driver side. > > We can not use the snd_pcm_lib_malloc_pages() in the pcm_new callback > because at this time the substream->runtime is not initialized and it > leads to the kernel crash. > > I used the apis as > > int snd_soc_pcm_new_wc_dma_buffer(struct snd_soc_pcm_runtime *rtd, > size_t max_bytes) > { > ::::::::::::: > substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; > if (substream) { > substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_WC; > ret = snd_pcm_lib_malloc_pages(substream, max_bytes); > if (ret) > goto err; > } > :::::::: > } > > > int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t > size) > { > struct snd_pcm_runtime *runtime; > struct snd_dma_buffer *dmab = NULL; > > if (PCM_RUNTIME_CHECK(substream)) > return -EINVAL; > > if (snd_BUG_ON(substream->dma_buffer.dev.type == > SNDRV_DMA_TYPE_UNKNOWN)) > return -EINVAL; > > runtime = substream->runtime; > > if (runtime->dma_buffer_p) { > ---------------Kernel crash at this point ---------- > > So I used the snd_dma_alloc_pages() from the driver to allocate WC memory. snd_pcm_lib_malloc_pages() should be called only from hw_params. Prior to that, you need to set up the dam_buffer type by calling snd_pcm_lib_preallocate_pages() or snd_pcm_lib_preallocate_pages_for_all() at the time to create a PCM instance. The size can be 0 if not necessary to allocate there but later on demand. > > > +/* allocate the coherent DMA pages */ > > +static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) > > +{ > > + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); > > This does not get compiled in ARM because dma_alloc_coherant is macro > defined as > > #define dma_alloc_coherent(d, s, h, f) dma_alloc_attrs(d, s, h, f, NULL) Ah, OK. thanks, Takashi > static inline void *dma_alloc_attrs(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag, > struct dma_attrs *attrs) > > > I fixed this by > static void *_dma_alloc_coherent(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag) > { > return dma_alloc_coherent(dev, size, dma_handle, flag); > } > > /* allocate the coherent DMA pages */ > static void *snd_malloc_dev_pages(struct device *dev, size_t size, > dma_addr_t *dma) > { > return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > } >

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Takashi Iwai 29 Jun '12

29 Jun '12

At Fri, 29 Jun 2012 15:53:15 +0530, Laxman Dewangan wrote: > > Some of the ARM based soc allocate the writecombine dma buffer for > pcm substreams. They have the same codes for managing this buffer. > Moving this to the core/pcm files so that they can use that directly. > > Remove the code from Tegra PCM and use these new library function. > > This is enabled only for ARM specific and can be extended to other > architecture if they support the writecombine dma buffer. > > This patch is based on detail discussion on patch: > [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap > And suggestion from Lars and Takashi. Looking through your patch, I think an easier integration is just to add writecombine option to memalloc.c which calls dma_alloc_writecombine() instead of dma_alloc_coherent(). The addition for mmap is still an open question. Again, an easier option so far looks like just add the call of dma_alloc_writecombine() in snd_pcm_lib_default_mmap(). Alternatively, we can create an individual mmap pcm_ops as we discussed. Below is a totally untested patch, but you can imagine what I meant. After this change, replace with snd_pcm_lib_malloc_pages with SNDRV_DMA_TYPE_DEV_WC, and that's all you need in the driver side. Takashi --- diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index c425062..7840340 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_WC 4 /* writecombine pages */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6915692..11108e8 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -124,8 +124,12 @@ void snd_free_pages(void *ptr, size_t size) */ #ifdef CONFIG_HAS_DMA -/* allocate the coherent DMA pages */ -static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +static void *__snd_malloc_dev_pages(struct device *dev, size_t size, + dma_addr_t *dma, + void *(*func)(struct device *dev, + size_t size, + dma_addr_t *dma_handle, + gfp_t flag)) { int pg; void *res; @@ -138,16 +142,18 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d | __GFP_COMP /* compound page lets parts be mapped */ | __GFP_NORETRY /* don't trigger OOM-killer */ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ - res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); + res = func(dev, PAGE_SIZE << pg, dma, gfp_flags); if (res != NULL) inc_snd_pages(pg); return res; } -/* free the coherent DMA pages */ -static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, - dma_addr_t dma) +static void __snd_free_dev_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma, + void (*func)(struct device *dev, + size_t size, void *vaddr, + dma_addr_t dma_handle)) { int pg; @@ -155,8 +161,36 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, return; pg = get_order(size); dec_snd_pages(pg); - dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); + func(dev, PAGE_SIZE << pg, ptr, dma); +} + +/* allocate the coherent DMA pages */ +static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +{ + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); +} + +/* free the coherent DMA pages */ +static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma) +{ + __snd_free_dev_pages(dev, ptr, dma, dma_free_coherent); } + +#ifdef CONFIG_ARM +/* allocate the writecombine DMA pages */ +static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, dma_addr_t *dma) +{ + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_writecombine); +} + +/* free the writecombine DMA pages */ +static void snd_free_dev_wc_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma) +{ + __snd_free_dev_pages(dev, ptr, dma, dma_free_writecombine); +} +#endif /* CONFIG_ARM */ #endif /* CONFIG_HAS_DMA */ /* @@ -200,6 +234,11 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; +#ifdef CONFIG_ARM + case SNDRV_DMA_TYPE_DEV_WC: + dmab->area = snd_malloc_dev_wc_pages(device, size, &dmab->addr); + break; +#endif #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: @@ -272,6 +311,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; +#ifdef CONFIG_ARM + case SNDRV_DMA_TYPE_DEV_WC: + snd_free_dev_wc_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); + break; +#endif #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: @@ -378,7 +422,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "WC" }; mutex_lock(&list_mutex); seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..c499791 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3170,6 +3170,14 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_RESERVED; +#ifdef CONFIG_ARM + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_WC) + return dma_mmap_writecombine(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV)

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC settings
by Prchal Jiří 29 Jun '12

29 Jun '12

This patch adds AGC target level and times settings for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 50a7112..5335ebc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -204,6 +204,10 @@ static const char *aic3x_right_hpcom_mux[] = static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -231,6 +235,21 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +#define LAGC_LEV_ENUM 0 +#define RAGC_LEV_ENUM 1 +#define LAGC_ATT_ENUM 2 +#define RAGC_ATT_ENUM 3 +#define LAGC_DEC_ENUM 4 +#define RAGC_DEC_ENUM 5 +static const struct soc_enum aic3x_agc_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -355,6 +374,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_agc_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_enum[RAGC_DEC_ENUM]), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, -- 1.7.7

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[alsa-devel] ASoC updates for 3.5
by Mark Brown 29 Jun '12

29 Jun '12

The following changes since commit 3419ae781f1592b3d367107db6500090495490cd: ASoC: tegra+wm8903: turn of mic detect when card is removed (2012-06-12 11:38:48 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to 5e531fc87875fc0bbfb3ca46a8a178674c4e9b7c: ASoC: tlv320aic3x: Fix codec pll configure bug (2012-06-27 12:46:25 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.5 A couple of unexciting driver specific changes plus a fix for gain controls used as DAPM controls which isn't yet needed in mainline but is a definite bug. ---------------------------------------------------------------- Benoît Thébaudeau (1): ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect Hebbar, Gururaja (1): ASoC: tlv320aic3x: Fix codec pll configure bug Mark Brown (1): ASoC: wm2200: Add missing BCLK rate sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/wm2200.c | 1 + sound/soc/soc-dapm.c | 8 +------- 4 files changed, 4 insertions(+), 10 deletions(-)

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[alsa-devel] [PATCH] ASoC: spdif: Build separate RX and TX objects
by Mark Brown 29 Jun '12

29 Jun '12

Otherwise we fail to link when building as modules due to multiple init/exit functions. Reported-by: Fengguang Wu <wfg(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 62c3d4d..acf8088 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,7 +45,8 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o +snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o @@ -159,7 +160,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o -obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o -- 1.7.10

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Re: [alsa-devel] [pulseaudio-discuss] [PATCH 2/4] alsa-mixer: Add Phantom Jack support
by Raymond Yau 29 Jun '12

29 Jun '12

2012/6/28 David Henningsson <david.henningsson(a)canonical.com>: > For kernel 3.6, "phantom jack" kctls have been added. They serve as > a marker that a particular port exist. They were made so we can detect > that there actually are speakers and internal mic on a laptop, even if > there are no other indications (volume controls etc). > > Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> > --- It seem that there will be numerous phantom ports created e.g. two internal mic phantom ports, spdif and spdif in, cd on the intel d975 desktop ./hda-emu codecs/stac9274d-intel-d975 CTRL: add: Front Headphone Jack:0 JACK created Front Headphone, type 1 JACK report Front Headphone, status 0 CTRL: add: Front Mic Jack:0 JACK created Front Mic, type 2 JACK report Front Mic, status 0 CTRL: add: Rear Mic Jack:0 JACK created Rear Mic, type 2 JACK report Rear Mic, status 0 CTRL: add: Internal Mic Phantom Jack:0 CTL Notify: Internal Mic Phantom Jack:0, mask=1 CTRL: add: Internal Mic Phantom Jack:1 CTL Notify: Internal Mic Phantom Jack:1, mask=1 CTRL: add: Line Jack:0 JACK created Line, type 32 JACK report Line, status 0 CTRL: add: CD Phantom Jack:0 CTL Notify: CD Phantom Jack:0, mask=1 CTRL: add: SPDIF Phantom Jack:0 CTL Notify: SPDIF Phantom Jack:0, mask=1 CTRL: add: SPDIF In Phantom Jack:0 CTL Notify: SPDIF In Phantom Jack:0, mask=1

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[alsa-devel] [PATCH] pcm: support for audio wallclock
by Pierre-Louis Bossart 28 Jun '12

28 Jun '12

add new snd_pcm_status_get_htstamp_wallclk() routine to query the audio timestamps read from the wallclock. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++-- src/pcm/pcm.c | 31 +++++++++++++++++++++++++++++++ 3 files changed, 38 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..542d9a8 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -550,6 +550,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -858,6 +859,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..f0d23d6 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..49d7b99 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3123,6 +3123,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6213,6 +6233,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio wallclock timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * -- 1.7.6.5

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 19:30:57 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 07:21 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 19:06:07 +0530, > > Laxman Dewangan wrote: > > > >>>> Then how about this? > >>>> create new file and header for snd-pcm-writecombine-buffer.c/.h and put > >>>> this in sound/core. > >>>> Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > >>>> which will be select this config through sound/soc/xxx/Kconfig if they > >>>> want to use. > >>>> This will provide three apis: new, free and mmap. > >>> Way too much hustles than necessary... > >> Ooh no. :-( I had taken the idea from snd_dmaengine_pcm driver. > >> Bit I like to hear simple way which solves the purpose. > > As Lars suggested, a simple ifdef should suffice for now. > > With the upcoming generic dma_mmap_writecombine() stuff, it'd be even > > simpler in future. > > > > But, still we need to be careful about this. As mentioned, there is > > no flag to know the possibility of writecombine mmap beforehand. > > It'd be nice if we have either a compile-time or a run-time flag / > > function to check that. Then the driver can also expose the mmap > > capability to user-space depending on the flag. > > Yaah, this seems really simple. Thanks for suggestion. > If I understand it fully, the new apis will be declare in sound/pcm.h, > implement it in pcm_native.c and use the config variable > CONFIG_SND_PCM_WRITECOMBINE_BUFFER for ifdef. Or this could be simply ifdef CONFIG_ARM. Since this ifdef will be unnecessary soon later, I don't think we need to give yet another kconfig. Of course, we should put a good comment around there for not keeping it for long time. Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Lars-Peter Clausen 28 Jun '12

28 Jun '12

On 06/28/2012 01:56 PM, Laxman Dewangan wrote: > On Thursday 28 June 2012 04:21 PM, Mark Brown wrote: >> * PGP Signed by an unknown key >> >> On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: >>> The pcm mmap for the substream dma buffer is done using the >>> dma_mmap_writecombine(). >>> Some of the chip sound driver like Spear, Tegra, Davinci, >>> ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this >>> mapping locally. >>> Add api in the snd dmaengine driver so that different client of >>> snd dmaengine can use this rather than implementing locally. >> OK, but I'd expect this to come along with one or more patches >> converting at least some of the users to the new API... > > This is effort towards moving the Tegra pcm driver for dmangine based dma > driver inplace of legacy dma driver for Tegra which have private API. > > Currently, following 4 pcm driver can use this API directly keeping that > they already using the generic snd dmaengine pcm driver. > ep93xx, > spear, > fsl, > mxs. > And Tegra will be next. > > Other driver which can use this API are davinci, OMAP, samsung but because > they are not using the snd_dmaengine_pcm driver's api as of now so not be > very much useful for them. Btw. all these drivers basically use the same code for buffer preallocation. Since how the buffer is mmaped is related to how it is allocated it might be worth to factor this out at the same time as well. - Lars

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 18:34:02 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 06:27 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 18:05:53 +0530, > > Laxman Dewangan wrote: > >> On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > >>> * PGP Signed by an unknown key > >>> > >>> On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >>>> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>>>> Oh, dear - this means we don't actually have a generic API at all. Is > >>>>> there any effort being made to make this generally available? > >>>> Yes there is, and in fact in next/master there is generic > >>>> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >>>> though. But in theory it should fallback to a normal coherent mapping. Marek > >>>> did a lot of work on this, I've put him on Cc. > >> So can we put the function snd_pcm_lib_writecombine_mmap() in the > >> pcm_native.c and only export this api for ARM i.e. under macro #ifdef > >> CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > >> directly use that one? > >> Or, wait for this common API until all ARCH support it? > > I think it's fine to put it first in ALSA side with some ifdef. > > A similar trick is already found for snd_pcm_lib_mmap_iomem. > > See include/sound/pcm.h. > > > > But, actually it's a still question what if an architecture doesn't > > support the mmap of writecombine at all. The proposed patch doesn't > > allow you to know whether writecombine-mmap is possible or not on the > > running architecture until you really try to call it and fail. > > It's a missing piece, IMO. > > Lars wanted to move the buffer allocation also to common place. Which common place? > Then how about this? > create new file and header for snd-pcm-writecombine-buffer.c/.h and put > this in sound/core. > Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > which will be select this config through sound/soc/xxx/Kconfig if they > want to use. > This will provide three apis: new, free and mmap. Way too much hustles than necessary... Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 18:05:53 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > > * PGP Signed by an unknown key > > > > On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>> Oh, dear - this means we don't actually have a generic API at all. Is > >>> there any effort being made to make this generally available? > >> Yes there is, and in fact in next/master there is generic > >> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >> though. But in theory it should fallback to a normal coherent mapping. Marek > >> did a lot of work on this, I've put him on Cc. > > So can we put the function snd_pcm_lib_writecombine_mmap() in the > pcm_native.c and only export this api for ARM i.e. under macro #ifdef > CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > directly use that one? > Or, wait for this common API until all ARCH support it? I think it's fine to put it first in ALSA side with some ifdef. A similar trick is already found for snd_pcm_lib_mmap_iomem. See include/sound/pcm.h. But, actually it's a still question what if an architecture doesn't support the mmap of writecombine at all. The proposed patch doesn't allow you to know whether writecombine-mmap is possible or not on the running architecture until you really try to call it and fail. It's a missing piece, IMO. Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Lars-Peter Clausen 28 Jun '12

28 Jun '12

On 06/28/2012 09:01 AM, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. > This is not really related to the dmaengine pcm driver. It's more of a coincidence that all upstream drivers which use the dmaengine pcm driver also use write-combined memory. In my opinion it would be better to add this to the ALSA core. Now that there is a generic dma_mmap_writecombine it may make sense to integrate this with snd_pcm_lib_default_mmap. - Lars > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > include/sound/dmaengine_pcm.h | 2 ++ > sound/soc/soc-dmaengine-pcm.c | 17 +++++++++++++++++ > 2 files changed, 19 insertions(+), 0 deletions(-) > > diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h > index b877334..15c0d3c 100644 > --- a/include/sound/dmaengine_pcm.h > +++ b/include/sound/dmaengine_pcm.h > @@ -47,4 +47,6 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream); > > struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); > > +int snd_dmaengine_pcm_mmap_writecombine(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma); > #endif > diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c > index 5df529e..6535eb3 100644 > --- a/sound/soc/soc-dmaengine-pcm.c > +++ b/sound/soc/soc-dmaengine-pcm.c > @@ -21,6 +21,7 @@ > #include <linux/module.h> > #include <linux/init.h> > #include <linux/dmaengine.h> > +#include <linux/dma-mapping.h> > #include <linux/slab.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -313,3 +314,19 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) > return 0; > } > EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); > + > +/** > + * snd_dmaengine_pcm_mmap_writecombine - mmap stream dma buffer using > + * dma_mmap_writecombine(). > + * @substream: PCM substream > + * @vma: VM area. > + */ > +int snd_dmaengine_pcm_mmap_writecombine(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + > + return dma_mmap_writecombine(substream->pcm->card->dev, vma, > + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); > +} > +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_mmap_writecombine);

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[alsa-devel] [PATCH] ASoC: dwc: Add missing __iomem annotations
by Mark Brown 28 Jun '12

28 Jun '12

Otherwise sparse gets very upset with us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e667e2b..1bd042b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -71,12 +71,12 @@ struct dw_i2s_dev { int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; -static inline void i2s_write_reg(void *io_base, int reg, u32 val) +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) { writel(val, io_base + reg); } -static inline u32 i2s_read_reg(void *io_base, int reg) +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) { return readl(io_base + reg); } -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 05:26:58PM +0530, Laxman Dewangan wrote: > Other driver which can use this API are davinci, OMAP, samsung but > because they are not using the snd_dmaengine_pcm driver's api as of > now so not be very much useful for them. There's no reason why we can't factor out more common code - this is essentially what Lars-Peter was saying. > Should I send one patch per driver change or squash all 4 file > changes in one patch? Whatever will be easy or recommended? It doesn't really matter that much, but please take on board Lars-Peter's comments.

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. OK, but I'd expect this to come along with one or more patches converting at least some of the users to the new API...

1 0

[alsa-devel] kcontrol get method
by raj raja 28 Jun '12

28 Jun '12

Hi, Generally get and set methods of Kcontrol called via mixer controls. I want to update variable value based on my kcontrol set method. I could not call get method, like as normal functions. So any way to call particular kcontrol get method in driver itself Thanks Raj

1 0

[alsa-devel] ALS100: no interrupts in Linux but works fine in DOS
by Ondrej Zary 27 Jun '12

27 Jun '12

Hello, I have a Freetek PRO32PNP card - an ISA card with ALS100 chip and onboard Crystal wavetable MIDI. The card works fine as SB16 in DOS - without any initialization utility, PnP BIOS configures the card to 0x220, irq5, dma1, dma5. However it does not work in Linux. Driver loads fine (snd-als100, even tried old OSS sb) but when I try to play anything, the player hangs without producing any sound. /proc/interrupts show that no interrupts are generated. Mixer works fine. The only suspicious thing is "unknown tag 0xff" and "invalid checksum" in dmesg (this is from patched kernel that shows size of the unknown tag): [ 0.159650] isapnp: Scanning for PnP cards... [ 0.246683] pnp 01:01.00: dependent set 0 (preferred) io min 0x220 max 0x220 align 0 size 16 flags 0x2 [ 0.246889] pnp 01:01.00: dependent set 0 (preferred) irq 5 flags 0x1 [ 0.247072] pnp 01:01.00: dependent set 0 (preferred) dma 5 (bitmask 0x20) flags 0x72 [ 0.247253] pnp 01:01.00: dependent set 0 (preferred) dma 1 (bitmask 0x2) flags 0x68 [ 0.247872] pnp 01:01.00: dependent set 1 (acceptable) io min 0x220 max 0x280 align 32 size 16 flags 0x0 [ 0.248078] pnp 01:01.00: dependent set 1 (acceptable) irq 5 7 9 10 flags 0x1 [ 0.248265] pnp 01:01.00: dependent set 1 (acceptable) dma 5 6 7 (bitmask 0xe0) flags 0x72 [ 0.248450] pnp 01:01.00: dependent set 1 (acceptable) dma 0 1 3 (bitmask 0xb) flags 0x68 [ 0.248963] pnp 01:01.00: dependent set 2 (functional) io min 0x100 max 0x3f0 align 16 size 16 flags 0x0 [ 0.249213] pnp 01:01.00: dependent set 2 (functional) irq 5 7 9 10 11 12 15 flags 0xf [ 0.249404] pnp 01:01.00: dependent set 2 (functional) dma 0 1 3 5 6 7 (bitmask 0xeb) flags 0x79 [ 0.249996] pnp 01:01.01: independent io min 0x388 max 0x388 align 0 size 8 flags 0x2 [ 0.250535] pnp 01:01.02: independent io min 0x200 max 0x200 align 0 size 2 flags 0x2 [ 0.251171] pnp 01:01.03: dependent set 0 (preferred) io min 0x330 max 0x330 align 0 size 2 flags 0x2 [ 0.251365] pnp 01:01.03: dependent set 0 (preferred) irq 9 flags 0x1 [ 0.251984] pnp 01:01.03: dependent set 1 (acceptable) io min 0x100 max 0x3f0 align 16 size 2 flags 0x0 [ 0.252244] pnp 01:01.03: dependent set 1 (acceptable) irq 5 7 9 10 11 12 15 flags 0xf [ 0.256256] pnp 01:01.03: unknown tag 0xff (card 1) size=18041, ignored [ 4.527093] 01:01: invalid checksum 0xad [ 4.527257] Switching to clocksource tsc [ 4.528594] 01:01: card 'ALS100 Media Audio Controller' [ 4.528699] isapnp: 1 Plug & Play card detected total I've even tried to disable Linux pnp completely (noisapnp pnpbios=off) and load snd-sb16 module manually (modprobe snd-sb16 isapnp=0) with exactly the same result (everything seems to be OK but there's no sound and interrupts). Any ideas? -- Ondrej Zary

1 0

[alsa-devel] periods_max initialization in snd_usb_hardware
by Amit Nagal 27 Jun '12

27 Jun '12

Hi , snd_usb_hardware is defined in sound/usb/pcm.c as : static struct snd_pcm_hardware snd_usb_hardware = { ..... .buffer_bytes_max = 1024 * 1024, .period_bytes_min = 64, .period_bytes_max = 512 * 1024, .periods_min = 2, .periods_max = 1024, }; here periods_max is intialized with value = 1024 . but going by buffer_bytes_max and period_bytes_min value defined above , periods_max = ( buffer_bytes_max / period_bytes_min ) = 16384 . why this upper cap of 1024 is maintained for periods_max even though from calculation it comes as 16384 ? Thanx & Regards Amit Nagal

2 1

[alsa-devel] how to update hw_ptr and appl_ptr
by Zhen Fu 27 Jun '12

27 Jun '12

Hi developer, I recently look the function snd_pcm_update_hw_ptr0 want to find how to update status->hw_ptr and control->appl_ptr. If you have to know this, can you tell me? Thank you very much. -- Best Regards, Zhen Fu

2 6

[alsa-devel] [PATCH 1/2] ALSA: HDA: Support single 3-pin jack without VREF on the actual pin
by David Henningsson 27 Jun '12

27 Jun '12

Some ASUS device has a single 3-pin jack that can either be a mic or a headphone, but the pin does not have VREF capabilities. We've been told by Realtek to instead enable VREF on pin 0x18 in that case. BugLink: https://bugs.launchpad.net/bugs/1018262 Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++------------- 1 file changed, 34 insertions(+), 15 deletions(-) Alsa info: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1018262/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5c81ee9..6174d4a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -303,6 +303,38 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) static void call_update_outputs(struct hda_codec *codec); static void alc_inv_dmic_sync(struct hda_codec *codec, bool force); +/* for shared I/O, change the pin-control accordingly */ +static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP/mic jack. + */ + + val = snd_hda_get_default_vref(codec, pin); + + /* This pin does not have vref caps - let's instead enable vref on pin 0x18 + instead, as suggested by Realtek */ + if (val == AC_PINCTL_VREF_HIZ) { + const hda_nid_t vref_pin = 0x18; + /* Sanity check pin 0x18 */ + if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN && + get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) { + unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); + if (vref_val != AC_PINCTL_VREF_HIZ) + snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0)); + } + } + + val = set_as_mic ? val | PIN_IN : PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); + + spec->automute_speaker = !set_as_mic; + call_update_outputs(codec); +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -329,21 +361,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; - /* for shared I/O, change the pin-control accordingly */ - if (spec->shared_mic_hp) { - unsigned int val; - hda_nid_t pin = spec->autocfg.inputs[1].pin; - /* NOTE: this assumes that there are only two inputs, the - * first is the real internal mic and the second is HP jack. - */ - if (spec->cur_mux[adc_idx]) - val = snd_hda_get_default_vref(codec, pin) | PIN_IN; - else - val = PIN_HP; - snd_hda_set_pin_ctl(codec, pin, val); - spec->automute_speaker = !spec->cur_mux[adc_idx]; - call_update_outputs(codec); - } + if (spec->shared_mic_hp) + update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); -- 1.7.9.5

2 3

Re: [alsa-devel] emu10k1 driver can't allocate memory
by Takashi Iwai 27 Jun '12

27 Jun '12

At Sun, 20 May 2012 03:04:14 +0200, Mikael Magnusson wrote: > > Hi, (please CC me on replies, not subscribed) > > I recently upgraded to 3.3.6 from 2.6.33.15 (yeah, I know). Since then > I noticed sometimes sounds were sometimes missing, then I noticed > thousands of these in dmesg, > [37280.283484] max memory size is 0x7fffffff (addr = 0x1d5d70071038000)!! I guess you are enabling IOMMU? I guess it translates the address somehow in a way emu10k1 doesn't like. There are really little changes in emu10k1 driver itself, so if the kernel upgrade changes the behavior, it must be in the core part such like IOMMU handling. Try to boot without iommu and see whether it changes the behavior again. Takashi > [37280.283486] emu: failure page = 32 > I also checked asfxload -M output, right after boot it shows all the > synth memory free as you would expect, but after a while of usage, it > is close to 0 (around 50KiB or so). Trying to run a simple program > that writes to /dev/audio fails with > write(3</dev/audio>, > "\32\21\t\3\0\0\4\n\23\34&/6:;94,#\31\17\10\2\0\0\4\v\24\36(07"..., > 4096) = -1 ENOMEM (Cannot allocate memory) > You can also note the output from asfxload -M dropping rapidly while > the program is making sound before it fails. Native alsa apps seem to > fare slightly better as the memory decrease caused by them is actually > returned when they close the device again. > > Loading and unloading soundfonts seem to work as normal, the memory is > returned when the soundfont is unloaded afaict. (I have the patch > applied that makes it possible to load soundfonts (changes the DMA32 > zone to 2GB). I don't think this is the cause of my problem since the > given addresses are way way outside the 4GB range as well.) > > Note that even well before the reported memory is at 0, apps randomly > fail to open the sound output and are silent. Curiously espeak when > successful doesn't cause a decrease in available memory, but when it > fails, it causes about 100KiB drop. > > I can't (don't dare) bisect because this is my production machine, but > I can test patches, or check the major releases between 2.6.33 and > 3.3.6 if you have no idea what causes this. > > I'm not the only one with the problem either, > http://www.google.com/search?nfpr=1&tbs=li:1&q=emu%3A+failure+page+%3D+32+m… > > -- > Mikael Magnusson >

2 3

[alsa-devel] Configuring an audio path within ALSA
by rashmi narayan 27 Jun '12

27 Jun '12

Dear ALSA Folks, I am a newbie to ALSA .I am trying to configure a path within the driver to open up /power up the path within the codec from MIC-IN to LineOUT. How can I do this within ALSA. Please send me some pointers. Thank you, Rashmi

3 6

[alsa-devel] [PATCH V3 1/1] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 27 Jun '12

27 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently(like its stretched). Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x changes in V3: use snd_soc_update_bits() instead of read-mask-write sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4f..e9b62b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -935,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - data | (pll_p << PLLP_SHIFT)); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb..08c7f66 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

2 1

[alsa-devel] [PATCH] ALSA: ASoC: Remove const attribute from lm49453's snd_soc_dai_driver
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following build warning. CC sound/soc/codecs/lm49453.o sound/soc/codecs/lm49453.c: In function ‘lm49453_i2c_probe’: sound/soc/codecs/lm49453.c:1511:24: warning: passing argument 3 of ‘snd_soc_register_codec’ discards qualifiers from pointer target type include/sound/soc.h:333:5: note: expected ‘struct snd_soc_dai_driver *’ but argument is of type ‘const struct snd_soc_dai_driver *’ Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f1..5bd1294 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1358,7 +1358,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { -- 1.7.10.2

3 6

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC, MIC_BIAS
by Prchal Jiří 27 Jun '12

27 Jun '12

This patch adds settings for TLV320AIC3106 of AGC, MIC_BIAS, De-emphasis. Also added default settings of output pop-up reduction, short circuit protection and mix both channels to mono_out to get maximum level. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -194,6 +194,11 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const char *aic3x_mic_bias[] = { "off", "2V", "2.5V", "AVDD" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -206,6 +211,13 @@ #define LINE2L_ENUM 8 #define LINE2R_ENUM 9 #define ADC_HPF_ENUM 10 +#define LAGC_LEV_ENUM 11 +#define RAGC_LEV_ENUM 12 +#define LAGC_ATT_ENUM 13 +#define RAGC_ATT_ENUM 14 +#define LAGC_DEC_ENUM 15 +#define RAGC_DEC_ENUM 16 +#define MIC_BIAS_ENUM 17 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -219,6 +231,13 @@ SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(MICBIAS_CTRL, 6, 4, aic3x_mic_bias), }; /* @@ -345,6 +364,15 @@ * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_enum[RAGC_DEC_ENUM]), + + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, @@ -352,6 +380,8 @@ SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + SOC_ENUM("Mic Bias", aic3x_enum[MIC_BIAS_ENUM]), }; /* @@ -1261,6 +1291,13 @@ snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + /* set to avoid artifacts on the audio output during power-on/off */ + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_B, 0x80); /*ac-coupled*/ + snd_soc_write(codec, HPOUT_POP_REDUCTION, 0x4e); /* 10 + 4 ms, reference*/ + + /* short circuit protection */ + snd_soc_write(codec, HPRCOM_CFG, 0x04); + /* DAC default volume and mute */ snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); @@ -1274,8 +1311,9 @@ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + /* mix both channels with -6dB level */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); /* unmute all outputs */ snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); @@ -1292,6 +1330,12 @@ /* By default route Line1 to ADC PGA mixer */ snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); + + /* AGC to -10dB, 20 / 500ms, no clip stepping, noise gate -90dB, hysteresis 3dB*/ + snd_soc_write(codec, LAGC_CTRL_A, 0x2f); + snd_soc_write(codec, RAGC_CTRL_A, 0x2f); + snd_soc_write(codec, LAGC_CTRL_C, 0xbe); + snd_soc_write(codec, RAGC_CTRL_C, 0xbe); /* PGA to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);

2 3

[alsa-devel] [PATCH V2 0/9] Add ASoC drivers for SPEAr platform & I2S driver for Designware module
by Rajeev Kumar 27 Jun '12

27 Jun '12

Changes since V1: 1. Move Synopsys i2s designware controller from 'spear' folder to a new folder 'dwc' 2. Implemented regmap in sta529 codec. 3. In-corporated review comments received on V1. Rajeev Kumar (6): sound:asoc: Add support for STA529 Audio Codec sound:asoc: Add support for synopsys i2s controller as per ASoC framework. sound:asoc: Add support for SPEAr ASoC pcm layer. sound:asoc: Add support for SPEAr ASoC machine driver. sound:asoc: Add Kconfig and Makefile to support SPEAr audio driver sound:asoc: Update Kconfig and Makefile(sound/soc/) to support SPEAr audio Vipin Kumar (3): sound:asoc: Add support for spdif in Audio Codec sound:asoc:spdif_in: Add spdif IN support sound:asoc:spdif_out: Add spdif out support include/sound/designware_i2s.h | 69 ++++++ include/sound/spear_dma.h | 35 +++ include/sound/spear_spdif.h | 29 +++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 76 ++++++ sound/soc/codecs/sta529.c | 477 +++++++++++++++++++++++++++++++++++++ sound/soc/dwc/Kconfig | 8 + sound/soc/dwc/Makefile | 3 + sound/soc/dwc/designware_i2s.c | 454 +++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 40 +++ sound/soc/spear/Makefile | 7 + sound/soc/spear/spdif_in.c | 297 +++++++++++++++++++++++ sound/soc/spear/spdif_in_regs.h | 60 +++++ sound/soc/spear/spdif_out.c | 397 ++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 ++++++ sound/soc/spear/spear_evb.c | 206 ++++++++++++++++ sound/soc/spear/spear_pcm.c | 214 +++++++++++++++++ 20 files changed, 2462 insertions(+), 1 deletions(-) create mode 100644 include/sound/designware_i2s.h create mode 100644 include/sound/spear_dma.h create mode 100644 include/sound/spear_spdif.h create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/dwc/Kconfig create mode 100644 sound/soc/dwc/Makefile create mode 100644 sound/soc/dwc/designware_i2s.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_evb.c create mode 100644 sound/soc/spear/spear_pcm.c

2 26

[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: initialize card pointer
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following warning: CC [M] sound/usb/caiaq/device.o sound/usb/caiaq/device.c: In function ‘snd_probe’: sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 64aed43..7da0d0a 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; - struct snd_card *card; + struct snd_card *card = NULL; struct usb_device *device = interface_to_usbdev(intf); ret = create_card(device, intf, &card); -- 1.7.10.2

2 2

[alsa-devel] [Regression] No sound on HP Pavilion dv6, STAC92xx
by Luis Henriques 27 Jun '12

27 Jun '12

Hi, After commit b0791dda813c179e539b0fc1ecd3f5f30f2571e2 ("ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops") we have several bug reports about not having sound on HP Pavilion laptops. Here's a link to one of these bug reports: http://bugs.launchpad.net/bugs/1013183 After reverting the referred commit, everything seems to be working fine again. Any ideas/fixes? Cheers, -- Luis

2 9

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds missing registers and bits for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -97,32 +97,20 @@ * There is no point in caching the reset register */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x00, 0x10, 0x04, 0x00, 0x00, 0x00, /* 0 */ + 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x80, /* 8 */ + 0x80, 0xff, 0xff, 0x78, 0x78, 0x78, 0x78, 0x78, /* 16 */ + 0x78, 0x00, 0x00, 0xfe, 0x00, 0x00, 0xfe, 0x00, /* 24 */ + 0x18, 0x18, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 32 */ + 0x00, 0x00, 0x00, 0x80, 0x80, 0x00, 0x00, 0x00, /* 40 */ + 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, /* 48 */ + 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* 56 */ + 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 64 */ + 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 72 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 80 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 88 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 96 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 104 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

3 3

Re: [alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

On Tue, Jun 26, 2012 at 14:31:42, Prchal Jiří wrote: > Hi Gururaja, > shouldn't be better to use: > > snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); > > instead of "read mask write" ? Sure will resend the patch. Thanks for the review > Even at this place you don't need to keep rest of register (PLL_Q and PLL_enable), so yust writing is OK. > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_p << PLLP_SHIFT); > > Dne 26.6.2012 08:03, Hebbar, Gururaja napsal(a): > > In sound/soc/codecs/tlv320aic3x.c > > > > data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > > > In the above code, pll-p value is OR'ed with previous value without > > clearing it. Bug is not seen if pll-p value doesn't change across > > Sampling frequency. > > > > However on some platforms (like AM335x EVM-SK), pll-p may have different > > values across different sampling frequencies. In such case, above code > > configures the pll with a wrong value. > > Because of this bug, when a audio stream is played with pll value > > different from previous stream, audio is heard as differently. > > > > Signed-off-by: Hebbar, Gururaja<gururaja.hebbar(a)ti.com> > > --- > > chnages in V2: > > modify subject to indicate ASOC and codec as tlv320aic3x > > > > sound/soc/codecs/tlv320aic3x.c | 2 +- > > sound/soc/codecs/tlv320aic3x.h | 1 + > > 2 files changed, 2 insertions(+), 1 deletions(-) > > > > diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c > > index d0dbac1..8c1977b 100644 > > --- a/sound/soc/codecs/tlv320aic3x.c > > +++ b/sound/soc/codecs/tlv320aic3x.c > > @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, > > } > > > > found: > > - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG)& ~PLLP_MASK; > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, > > diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h > > index 06a1978..16d9999 100644 > > --- a/sound/soc/codecs/tlv320aic3x.h > > +++ b/sound/soc/codecs/tlv320aic3x.h > > @@ -166,6 +166,7 @@ > > > > /* PLL registers bitfields */ > > #define PLLP_SHIFT 0 > > +#define PLLP_MASK 7 > > #define PLLQ_SHIFT 3 > > #define PLLR_SHIFT 0 > > #define PLLJ_SHIFT 2 > > Regards, Gururaja

1 0

[alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add input clock selection
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds input selection of main codec clock - from what pin. Tested with TLV320AIC3106 on BCLK and MCLK. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c @@ -972,6 +972,10 @@ struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, clk_id); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, clk_id); + aic3x->sysclk = freq; return 0; } --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h @@ -178,6 +178,13 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define CLKDIV_IN_MASK 0xc0 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 + /* Software reset register bits */ #define SOFT_RESET 0x80

2 2

[alsa-devel] [PATCH] ASoC: wm8994: Don't suspend accessory detection
by Mark Brown 26 Jun '12

26 Jun '12

Leave it up to the machine driver to disable accessory detection if desired, the common pattern is to have accessory detection be a wake source. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 37 ------------------------------------- 1 file changed, 37 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6474bae..a4aafdb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2967,23 +2967,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - switch (control->type) { - case WM8994: - snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); - break; - case WM1811: - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, 0); - /* Fall through */ - case WM8958: - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, 0); - break; - } - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -3033,28 +3018,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) i + 1, ret); } - switch (control->type) { - case WM8994: - if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) - snd_soc_update_bits(codec, WM8994_MICBIAS, - WM8994_MICD_ENA, WM8994_MICD_ENA); - break; - case WM1811: - if (wm8994->jackdet && wm8994->jack_cb) { - /* Restart from idle */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_JACK); - break; - } - break; - case WM8958: - if (wm8994->jack_cb) - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); - break; - } - return 0; } #else -- 1.7.10

1 0

[alsa-devel] [PATCH] davinci:mcasp: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as stretched. Fix the same by clearing the mask and then writing the value. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Add Realtek ALC280 codec support
by David Henningsson 26 Jun '12

26 Jun '12

This chip looks very similar to ALC269 and ALC27* variants. The bug reporter has verified that sound was working after this patch had been applied. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1017017 Tested-by: Richard Crossley <richardcrossley(a)o2.co.uk> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) Alsa info available at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1017017/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 41475ae..a5534b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6800,6 +6800,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, + { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- 1.7.9.5

2 1

Re: [alsa-devel] Using UCM with PulseAudio
by Arun Raghavan 25 Jun '12

25 Jun '12

On Thu, 2012-06-21 at 16:30 +0800, Feng Wei wrote: > 2012/6/20 Arun Raghavan <arun.raghavan(a)collabora.co.uk>: > > Liam, > > Thanks for clearing things up. > > > > On Fri, 2012-06-15 at 18:08 +0100, Liam Girdwood wrote: > > [...] > >> > The first problem is mutual exclusivity of verbs. From what I can > >> > understand, verbs are intended to be mutually exclusive -- if you have a > >> > HiFi verb and a VoiceCall verb, only one may be used at a time. We have > >> > mapped verbs to card profiles, which offer the same guarantee. However, > >> > on Android (which is a fair example of the kind of audio policy we might > >> > want), the HiFi verb PCMs maybe used while the VoiceCall PCMs are open. > >> > This is done, for example, to play an end-of-call tone from the CPU > >> > while the modem PCMs are still held open. Is there some way to do this > >> > with UCM? > >> > > >> > >> A verb is not tied to any specific PCM device here. It's intended to be > >> the highest level of audio use case where it can configure any audio > >> resource (including multiple cards) to enable the use case action. > >> > >> So for OMAP4 we would have the HiFi verb for all use case where we are > >> playing or capturing HiFi quality and the voice call verb where we are > >> making a telephone voice call. > >> > >> UCM provides the "modifier" to allow ad-hoc modifications to the audio > >> use case like above where we want to play an end of call tone. In this > >> case the verb is still VoiceCall, but PA would enable the "PlayTone" > >> modifer to play the tone (UCM can also tell Pulseaudio the sink PCM and > >> volume control for the tone data). > > > > Just as an observation, I think the Android HAL just uses PCM 0 and not > > the PCM 3, but in the PA case, I'll do it the way you describe since > > that makes more sense. > > > >> > The second problem is having separate PCMs for modifiers. In the OMAP4 > >> > profile, ringtone playback is exposed via a PlayTone modifier which > >> > corresponds to a separate PCM from regular HiFi playback. In the UCM-PA > >> > mapping we decided on, modifiers were implemented as device intended > >> > roles on a sink, so that when a stream with that role came in, we could > >> > enable the modifier, and disable it when such a stream ends. However, > >> > this doesn't account for switching the PCM on which playback is > >> > occurring. Should we be creating a separate sink for such modifiers > >> > (with lower priority, so they're not routed to unless there's a stream > >> > with the required role coming in)? Or should we be reopening the PCM for > >> > this? > >> > >> The intention for modifiers that use separate ALSA PCM sinks/sources > >> from the verb is to keep the main stream on the verb PCM source/sink and > >> the modifier stream will use the modifier PCM sink/source (this can be > >> the same PCM for some hardware). > >> > >> e.g. MP3 will be played to pcm 0 sink and ringtone to pcm 1 sink. The HW > >> will then mix both streams before they are rendered. > > > > Okay, so what this means is that we need to extend the current UCM work > > to create a second, lower priority sink for each modifier that has a > > distinct PlaybackPCM, and let the role-based routing pick that in cases > > where it makes sense. It's good that this doesn't change how we've > > mapped existing concepts -- just adds to it. > Sounds fine to me. The only problem is can we always open the modifier > pcm when modifier is not enabled. If not, I prefer to create/destroy > the sink by events. The alternative is to always add the sink and enable/disable the modifier in the I/O thread since we will get a message when a sink-input is added/removed. I'm just hacking this bit up, and will post patches if it all works out. There is another question here -- whether the modifier sink should have any ports or not. If I understand this correctly, we could let the actual routing be handled by the "main" sink's port and have the modifier sinks not touch that at all by not having any ports. In the ideal world though, it would be nice to have the ability to mark ports as being "dependent" somehow, so that port changes on the main sink would reflect in all others. This would be particularly useful for sinks and sources that have correlated paths like they do in the OMAP4 case. -- Arun

1 1

Re: [alsa-devel] Possible message repeat
by Clemens Ladisch 25 Jun '12

25 Jun '12

Brent Weatherall wrote: > our goal is to specify a special XU that WOULD require vendors > supporting it to modify their firmware. We fully expect a vendor to > modify their firmware to support our specification. That being said, > is there a simple way to use ALSA to communicate with a vendor > specific XU as long as the software understands the XU? For example, > in the video4linux2 world, we can specify a UVC XU with a set of > controls for special processing, then vendors that want to support it > modify their firmware to support it. Then anyone can use our custom > user-space drivers to easily talk to the video device. V4L2 has > controls made for using XUs in this manner. The audio class specification does not define any mechanism for a device to specify what those controls would be. And because nobody has ever needed to use XUs so far, the current ALSA driver has no other way to create such controls. If you insist on not changing ALSA, you must bypass it. Regards, Clemens

2 1

[alsa-devel] wm8994 trivial merge fix
by Takashi Iwai 25 Jun '12

25 Jun '12

FYI, the patch below is what I applied from Jaroslav's tree. I simply picked it up directly into my tree (fix/asoc and for-linus branches) because of triviality. Takashi --- From 6cab3e1e65bfe070f09bb091eeda182b171d5929 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Mon, 28 May 2012 11:14:33 +0200 Subject: [PATCH] ASoC: wm8994: remove duplicate code It seems that the code duplication was added at a merge operation. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm8994.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index aa8c98b..1436b6c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -727,9 +727,6 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (!wm8994->jackdet || !wm8994->jack_cb) return; - if (!wm8994->jackdet || !wm8994->jack_cb) - return; - if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: dwc: Bodge around continuing absence of clock API stubs
by Mark Brown 25 Jun '12

25 Jun '12

The patches for stubbing out the generic clock API still haven't been applied so we need to either add ifdefs here or add a dependency until someone decides to actually apply them. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index 93e9fc3..e334900 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,5 +1,6 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" + depends on CLKDEV_LOOKUP help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto -- 1.7.10

1 0

[alsa-devel] [PATCH] ASoC: dmaengine_pcm: fix typo in comment
by Uwe Kleine-König 25 Jun '12

25 Jun '12

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/soc-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 4756952..5647449 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -243,7 +243,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd * Note that this function will use private_data field of the substream's * runtime. So it is not availabe to your pcm driver implementation. If you need * to keep additional data attached to a substream use - * snd_dmaeinge_pcm_{set,get}_data. + * snd_dmaengine_pcm_{set,get}_data. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- 1.7.10

2 1

[alsa-devel] [RFC PATCH] Inverted internal mic
by David Henningsson 25 Jun '12

25 Jun '12

Hi, One of the more common problems on laptop machines, is that the internal mic provides a stereo signal but with one channel phase inverted, or differential output. This means problems for applications recording two channels but later merging them into one, leaving them with zero or near-zero output. There are various ways we can work around this in both the kernel, alsa-lib and PulseAudio layers. It's a matter of picking the right one. I'm leaning towards trying to fix it in the kernel's codec drivers, because 1) we already have quirking infrastructure there 2) we already have some working quirks already in that layer 3) it would benefit other sound applications that use ALSA directly. The downside to that is really that we're silencing out one channel for everyone, leading to no application being able to use both channels, even if they would implement some kind of "auto-detect-and-reverse-one-channel" functionality. For the most part, this has been some Acer Aspire machines running Realtek ALC268 / ALC269 / ALC271X / ALC272X, and for two of these we have proc coefs we can set, but for the other two these proc coefs, if they exist, are unknown to us. Recently I came across a Conexant as well, and I decided to write a patch for it, that would take the approach that the internal mic is forced mono on the kcontrol, and make sure the right channel is always muted. The patch is verified by the reporter to fix this problem. It could use some perfection though - it would make sense to to the same to the internal mic boost as well, and the strcmp('Internal Mic') call could maybe be turned into something more elegant. But before going ahead with that, I'd like to hear your opinion on the matter, if you agree that this is a good approach to the problem? References: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/903853/+attachme… https://bugs.launchpad.net/bugs/903853 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

5 38

[alsa-devel] Exclusive access to capture device
by Guy Rutenberg 24 Jun '12

24 Jun '12

Hi, Is there a way to open a capture device exclusively? E.g. I want to capture audio and be sure no other application can capture from that device at the same time. I've went through the API and searched the mailing list and haven't found a way to do so. Thanks a lot, Guy ---- http://www.guyrutenberg.com/

2 3

[alsa-devel] [PATCH 1/2] ASoC: Add shared code for Wolfson Arizona class devices
by Mark Brown 23 Jun '12

23 Jun '12

The Wolfson Arizona series of audio hub CODECs can share a large amount of their driver code as the result of a common register map. This patch adds some of this core support, providing a basis for the initial WM5102 audio driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 149 +++++++++ 4 files changed, 937 insertions(+) create mode 100644 sound/soc/codecs/arizona.c create mode 100644 sound/soc/codecs/arizona.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 43f5240..2ae8082 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,6 +129,11 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate +config SND_SOC_ARIZONA + tristate + default y if SND_SOC_WM5102=y + default m if SND_SOC_WM5102=m + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d30654..3005ea6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l52-objs := cs42l52.o @@ -128,6 +129,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o +obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c new file mode 100644 index 0000000..3b5730b --- /dev/null +++ b/sound/soc/codecs/arizona.c @@ -0,0 +1,781 @@ +/* + * arizona.c - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gcd.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" + +#define ARIZONA_AIF_BCLK_CTRL 0x00 +#define ARIZONA_AIF_TX_PIN_CTRL 0x01 +#define ARIZONA_AIF_RX_PIN_CTRL 0x02 +#define ARIZONA_AIF_RATE_CTRL 0x03 +#define ARIZONA_AIF_FORMAT 0x04 +#define ARIZONA_AIF_TX_BCLK_RATE 0x05 +#define ARIZONA_AIF_RX_BCLK_RATE 0x06 +#define ARIZONA_AIF_FRAME_CTRL_1 0x07 +#define ARIZONA_AIF_FRAME_CTRL_2 0x08 +#define ARIZONA_AIF_FRAME_CTRL_3 0x09 +#define ARIZONA_AIF_FRAME_CTRL_4 0x0A +#define ARIZONA_AIF_FRAME_CTRL_5 0x0B +#define ARIZONA_AIF_FRAME_CTRL_6 0x0C +#define ARIZONA_AIF_FRAME_CTRL_7 0x0D +#define ARIZONA_AIF_FRAME_CTRL_8 0x0E +#define ARIZONA_AIF_FRAME_CTRL_9 0x0F +#define ARIZONA_AIF_FRAME_CTRL_10 0x10 +#define ARIZONA_AIF_FRAME_CTRL_11 0x11 +#define ARIZONA_AIF_FRAME_CTRL_12 0x12 +#define ARIZONA_AIF_FRAME_CTRL_13 0x13 +#define ARIZONA_AIF_FRAME_CTRL_14 0x14 +#define ARIZONA_AIF_FRAME_CTRL_15 0x15 +#define ARIZONA_AIF_FRAME_CTRL_16 0x16 +#define ARIZONA_AIF_FRAME_CTRL_17 0x17 +#define ARIZONA_AIF_FRAME_CTRL_18 0x18 +#define ARIZONA_AIF_TX_ENABLES 0x19 +#define ARIZONA_AIF_RX_ENABLES 0x1A +#define ARIZONA_AIF_FORCE_WRITE 0x1B + +#define arizona_fll_err(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_warn(_fll, fmt, ...) \ + dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_dbg(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + +#define arizona_aif_err(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_warn(_dai, fmt, ...) \ + dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_dbg(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + +const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "Haptics", + "AEC", + "Mic Mute Mixer", + "Noise Generator", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "SLIMRX1", + "SLIMRX2", + "SLIMRX3", + "SLIMRX4", + "SLIMRX5", + "SLIMRX6", + "SLIMRX7", + "SLIMRX8", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "DRC2L", + "DRC2R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", +}; +EXPORT_SYMBOL_GPL(arizona_mixer_texts); + +int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { + 0x00, /* None */ + 0x04, /* Tone */ + 0x05, + 0x06, /* Haptics */ + 0x08, /* AEC */ + 0x0c, /* Noise mixer */ + 0x0d, /* Comfort noise */ + 0x10, /* IN1L */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x20, /* AIF1RX1 */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, /* AIF2RX1 */ + 0x29, + 0x30, /* AIF3RX1 */ + 0x31, + 0x38, /* SLIMRX1 */ + 0x39, + 0x3a, + 0x3b, + 0x3c, + 0x3d, + 0x3e, + 0x3f, + 0x50, /* EQ1 */ + 0x51, + 0x52, + 0x53, + 0x58, /* DRC1L */ + 0x59, + 0x5a, + 0x5b, + 0x60, /* LHPF1 */ + 0x61, + 0x62, + 0x63, + 0x68, /* DSP1.1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x90, /* ASRC1L */ + 0x91, + 0x92, + 0x93, +}; +EXPORT_SYMBOL_GPL(arizona_mixer_values); + +const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); +EXPORT_SYMBOL_GPL(arizona_mixer_tlv); + +static const char *arizona_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +const struct soc_enum arizona_lhpf1_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); + +const struct soc_enum arizona_lhpf2_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); + +const struct soc_enum arizona_lhpf3_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); + +const struct soc_enum arizona_lhpf4_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); + +int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_in_ev); + +int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_out_ev); + +int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + char *name; + unsigned int reg; + unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK; + unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT; + unsigned int *clk; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + name = "SYSCLK"; + reg = ARIZONA_SYSTEM_CLOCK_1; + clk = &priv->sysclk; + mask |= ARIZONA_SYSCLK_FRAC; + break; + case ARIZONA_CLK_ASYNCCLK: + name = "ASYNCCLK"; + reg = ARIZONA_ASYNC_CLOCK_1; + clk = &priv->asyncclk; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + break; + case 11289600: + case 12288000: + val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 22579200: + case 24576000: + val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 45158400: + case 49152000: + val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + default: + return -EINVAL; + } + + *clk = freq; + + if (freq % 6144000) + val |= ARIZONA_SYSCLK_FRAC; + + dev_dbg(arizona->dev, "%s set to %uHz", name, freq); + + return regmap_update_bits(arizona->regmap, reg, mask, val); +} +EXPORT_SYMBOL_GPL(arizona_set_sysclk); + +static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mode, base; + + base = dai->driver->base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mode = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + break; + case SND_SOC_DAIFMT_I2S: + mode = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = 3; + break; + default: + arizona_aif_err(dai, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + default: + arizona_aif_err(dai, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= ARIZONA_AIF1_BCLK_INV; + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= ARIZONA_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); + + return 0; +} + +static const int arizona_48k_rates[] = { + -1, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static const int arizona_44k1_rates[] = { + -1, + 44100, + 58800, + 88200, + 117600, + 177640, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579200, +}; + +static int arizona_sr_vals[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int arizona_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + const int *rates; + int i; + int bclk, lrclk, wl, frame, sr_val; + + if (params_rate(params) % 8000) + rates = &arizona_44k1_rates[0]; + else + rates = &arizona_48k_rates[0]; + + for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) { + if (rates[i] == snd_soc_params_to_bclk(params)) { + bclk = i; + break; + } + } + if (i == ARRAY_SIZE(arizona_44k1_rates)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for the chip. + */ + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + + arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", + rates[bclk], rates[bclk] / lrclk); + + wl = snd_pcm_format_width(params_format(params)); + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + + return 0; +} + +const struct snd_soc_dai_ops arizona_dai_ops = { + .set_fmt = arizona_set_fmt, + .hw_params = arizona_hw_params, +}; + +static irqreturn_t arizona_fll_lock(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "Locked\n"); + + complete(&fll->lock); + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_fll_clock_ok(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "clock OK\n"); + + complete(&fll->ok); + + return IRQ_HANDLED; +} + +static struct { + unsigned int min; + unsigned int max; + u16 fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +struct arizona_fll_cfg { + int n; + int theta; + int lambda; + int refdiv; + int outdiv; + int fratio; +}; + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + + /* Fref must be <=13.5MHz */ + div = 1; + cfg->refdiv = 0; + while ((Fref / div) > 13500000) { + div *= 2; + cfg->refdiv++; + + if (div > 8) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 1; + while (Fout * div < 90000000) { + div++; + if (div > 7) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + cfg->fratio = fll_fratios[i].fratio; + ratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return -EINVAL; + } + + cfg->n = target / (ratio * Fref); + + if (target % Fref) { + gcd_fll = gcd(target, ratio * Fref); + arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); + + cfg->theta = (target - (cfg->n * ratio * Fref)) + / gcd_fll; + cfg->lambda = (ratio * Fref) / gcd_fll; + } else { + cfg->theta = 0; + cfg->lambda = 0; + } + + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + cfg->n, cfg->theta, cfg->lambda); + arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", + cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + + return 0; + +} + +static void arizona_apply_fll(struct arizona *arizona, unsigned int base, + struct arizona_fll_cfg *cfg, int source) +{ + regmap_update_bits(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + + regmap_update_bits(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); +} + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona *arizona = fll->arizona; + struct arizona_fll_cfg cfg, sync; + unsigned int reg, val; + int syncsrc; + bool ena; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, &reg); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + ena = reg & ARIZONA_FLL1_ENA; + + if (Fout) { + /* Do we have a 32kHz reference? */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + syncsrc = -1; + } + + if (source == syncsrc) + syncsrc = -1; + + if (syncsrc >= 0) { + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + if (ret != 0) + return ret; + } else { + ret = arizona_calc_fll(fll, &cfg, Fref, Fout); + if (ret != 0) + return ret; + } + } else { + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (ena) + pm_runtime_put_autosuspend(arizona->dev); + + return 0; + } + + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + if (syncsrc >= 0) { + arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); + arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); + } else { + arizona_apply_fll(arizona, fll->base, &cfg, source); + } + + if (!ena) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (syncsrc >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(25)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll); + +int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, + int ok_irq, struct arizona_fll *fll) +{ + int ret; + + init_completion(&fll->lock); + init_completion(&fll->ok); + + fll->id = id; + fll->base = base; + fll->arizona = arizona; + + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); + snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), + "FLL%d clock OK", id); + + ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, + arizona_fll_lock, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", + id, ret); + } + + ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, + arizona_fll_clock_ok, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", + id, ret); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_fll); + +MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h new file mode 100644 index 0000000..8c2ca1d --- /dev/null +++ b/sound/soc/codecs/arizona.h @@ -0,0 +1,149 @@ +/* + * arizona.h - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ASOC_ARIZONA_H +#define _ASOC_ARIZONA_H + +#include <linux/completion.h> + +#include <sound/soc.h> + +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 + +#define ARIZONA_CLK_SRC_MCLK1 0x0 +#define ARIZONA_CLK_SRC_MCLK2 0x1 +#define ARIZONA_CLK_SRC_FLL1 0x4 +#define ARIZONA_CLK_SRC_FLL2 0x5 +#define ARIZONA_CLK_SRC_AIF1BCLK 0x8 +#define ARIZONA_CLK_SRC_AIF2BCLK 0x9 +#define ARIZONA_CLK_SRC_AIF3BCLK 0xa + +#define ARIZONA_FLL_SRC_MCLK1 0 +#define ARIZONA_FLL_SRC_MCLK2 1 +#define ARIZONA_FLL_SRC_SLIMCLK 2 +#define ARIZONA_FLL_SRC_FLL1 3 +#define ARIZONA_FLL_SRC_FLL2 4 +#define ARIZONA_FLL_SRC_AIF1BCLK 5 +#define ARIZONA_FLL_SRC_AIF2BCLK 6 +#define ARIZONA_FLL_SRC_AIF3BCLK 7 +#define ARIZONA_FLL_SRC_AIF1LRCLK 8 +#define ARIZONA_FLL_SRC_AIF2LRCLK 9 +#define ARIZONA_FLL_SRC_AIF3LRCLK 10 + +#define ARIZONA_MIXER_VOL_MASK 0x00FE +#define ARIZONA_MIXER_VOL_SHIFT 1 +#define ARIZONA_MIXER_VOL_WIDTH 7 + +struct arizona; + +struct arizona_priv { + struct arizona *arizona; + int sysclk; + int asyncclk; +}; + +#define ARIZONA_NUM_MIXER_INPUTS 55 + +extern const unsigned int arizona_mixer_tlv[]; +extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; + +#define ARIZONA_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + +#define ARIZONA_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + arizona_mixer_texts, arizona_mixer_values) + +#define ARIZONA_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define ARIZONA_MIXER_ENUMS(name, base_reg) \ + static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static ARIZONA_MUX_CTL_DECL(name##_in1); \ + static ARIZONA_MUX_CTL_DECL(name##_in2); \ + static ARIZONA_MUX_CTL_DECL(name##_in3); \ + static ARIZONA_MUX_CTL_DECL(name##_in4) + +#define ARIZONA_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define ARIZONA_MIXER_WIDGETS(name, name_str) \ + ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \ + ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \ + ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \ + ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define ARIZONA_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") + +extern const struct soc_enum arizona_lhpf1_mode; +extern const struct soc_enum arizona_lhpf2_mode; +extern const struct soc_enum arizona_lhpf3_mode; +extern const struct soc_enum arizona_lhpf4_mode; + +extern int arizona_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); +extern int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); + +extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); + +extern const struct snd_soc_dai_ops arizona_dai_ops; + +#define ARIZONA_FLL_NAME_LEN 20 + +struct arizona_fll { + struct arizona *arizona; + int id; + unsigned int base; + struct completion lock; + struct completion ok; + + char lock_name[ARIZONA_FLL_NAME_LEN]; + char clock_ok_name[ARIZONA_FLL_NAME_LEN]; +}; + +extern int arizona_init_fll(struct arizona *arizona, int id, int base, + int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); + +#endif -- 1.7.10

1 1

Re: [alsa-devel] [PATCH] sound: fix MODULE_LICENSE typos
by Mark Brown 23 Jun '12

23 Jun '12

On Fri, Jun 22, 2012 at 04:21:49PM -0700, Kees Cook wrote: > include/linux/license.h does not recognize "GPLv2", so adjust these > declarations to add the needed space before the version. This patch isn't against current -next which doesn't have this issue, I've no idea what you generated against as a version with this has never even been in -next. Please do also use subject lines that resemble those used in the rest of the subsystem, having to hand edit commit logs is tedious. It seems more sensible to fix license.h, GPLv2 is a pretty common way of writing the license name in the wider community.

1 0

[alsa-devel] [PATCH] ASoC: core: Add DOUBLE_R variants of the _RANGE controls
by Mark Brown 22 Jun '12

22 Jun '12

The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e4348d2..e063380 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -42,6 +42,10 @@ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ .max = xmax, .platform_max = xmax, .invert = xinvert}) +#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ + .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -96,6 +100,13 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -114,6 +125,16 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: dapm: Try to add all routes even if one fails
by Mark Brown 22 Jun '12

22 Jun '12

We may as well print as many errors as we can in one go rather than requiring developers to iterate through all their typos. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 667a941..3e7baab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2276,15 +2276,15 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i, r, ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(dapm, route); - if (ret < 0) { + r = snd_soc_dapm_add_route(dapm, route); + if (r < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - break; + ret = r; } route++; } -- 1.7.10

2 1

[PATCH] ALSA: hda - Add inverted mic quirks for Asus U41SV, Acer 1810TZ and AOD260
by David Henningsson 22 Jun '12

22 Jun '12

These machines have inverted phase on right channel for their internal mics. BugLink: https://bugs.launchpad.net/bugs/997227 BugLink: https://bugs.launchpad.net/bugs/996611 BugLink: https://bugs.launchpad.net/bugs/1006089 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53cd730..9be7da1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5918,6 +5918,14 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, } } +static void alc269_fixup_inv_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + alc_add_inv_dmic_mixer(codec, 0x12); +} + + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5936,6 +5944,7 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_MIC2_MUTE_LED, + ALC269_FIXUP_INV_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -6060,12 +6069,19 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, }, + [ALC269_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_inv_dmic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -6534,12 +6550,7 @@ static void alc272_fixup_mario(struct hda_codec *codec, "hda_codec: failed to override amp caps for NID 0x2\n"); } -static void alc662_fixup_inv_dmic(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - if (action == ALC_FIXUP_ACT_PROBE) - alc_add_inv_dmic_mixer(codec, 0x12); -} +#define alc662_fixup_inv_dmic alc269_fixup_inv_dmic enum { ALC662_FIXUP_ASPIRE, -- 1.7.9.5 --------------030106040506020209090705--

1 0

[alsa-devel] [PATCH] ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x
by Mark Brown 22 Jun '12

22 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da732x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 04af369..01be2a3 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1543,7 +1543,7 @@ static int da732x_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver soc_codec_dev_da732x = { +static struct snd_soc_codec_driver soc_codec_dev_da732x = { .probe = da732x_probe, .remove = da732x_remove, .set_bias_level = da732x_set_bias_level, -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: isabelle: Staticise non-exported isabelle_dai
by Mark Brown 22 Jun '12

22 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/isabelle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 0d62f3b..5d8f39e 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1036,7 +1036,7 @@ static struct snd_soc_dai_ops isabelle_ul_dai_ops = { }; /* ISABELLE dai structure */ -struct snd_soc_dai_driver isabelle_dai[] = { +static struct snd_soc_dai_driver isabelle_dai[] = { { .name = "isabelle-dl1", .playback = { -- 1.7.10

1 0

[alsa-devel] [PATCH] ALSA: hda - Don't power up when not powered down.
by Dylan Reid 22 Jun '12

22 Jun '12

After cancel_delayed_work_sync returns, the power down work either never started (power_on == 1) or finished (power_on == 0). In the former case there is no need to power up again. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 604699c..045b5e7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4444,6 +4444,13 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); + /* If the power down delayed work was cancelled above before starting, + * then there is no need to go through power up here. + */ + if (codec->power_on) { + spin_unlock(&codec->power_lock); + return; + } trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; -- 1.7.9.rc0

2 3

[alsa-devel] Need ALSA soc architecture Documents
by raj raja 22 Jun '12

22 Jun '12

Hi, I am very new to ALSA soc core . I want to have some basic knowledge on ALSA soc core architecture Where can i find the related pfd or ppt format documents Thanks Yuvaraj R

1 0

[alsa-devel] [PATCH V3 12/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 22 Jun '12

22 Jun '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Clemens Ladisch <clemens(a)ladisch.de> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 + sound/pci/hda/patch_conexant.c | 52 ++++++++++++++++++++++++++++++++++++++- 2 files changed, 52 insertions(+), 2 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ab741b0..d8b0a52 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2906,3 +2906,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172370b..ca3cc237 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int lemote:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; @@ -2278,7 +2279,7 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); - else if (spec->asus) + else if (spec->asus || spec->lemote) cxt5066_asus_automic(codec); } @@ -2911,6 +2912,32 @@ static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_lemote[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1: Connection index: 0 */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* ADC2, ADC3: unused */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2948,6 +2975,8 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_LEMOTE_A1004, /* Lemote Laptop A1004 */ + CXT5066_LEMOTE_A1205, /* Lemote All-In-One A1205 */ CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -2961,6 +2990,8 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_LEMOTE_A1004] = "lemote-laptop-a1004", + [CXT5066_LEMOTE_A1205] = "lemote-aio-a1205", [CXT5066_AUTO] = "auto", }; @@ -2993,6 +3024,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2011, "Lemote A1004", CXT5066_LEMOTE_A1004), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2012, "Lemote A1205", CXT5066_LEMOTE_A1205), SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3073,7 +3106,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->mic_boost = 3; /* default 30dB gain */ break; - + case CXT5066_LEMOTE_A1004: + case CXT5066_LEMOTE_A1205: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_lemote; + spec->num_init_verbs++; + spec->lemote = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.7.3

1 0

[alsa-devel] [PATCH] asoc: bfin: use dev_err to print error log instead of dev_dbg
by Scott Jiang 21 Jun '12

21 Jun '12

Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf6xx-sport.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index f19a72b..318c5ba 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -256,9 +256,9 @@ static irqreturn_t sport_err_irq(int irq, void *dev_id) struct device *dev = &sport->pdev->dev; if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) - dev_dbg(dev, "sport error: TUVF\n"); + dev_err(dev, "sport error: TUVF\n"); if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) - dev_dbg(dev, "sport error: ROVF\n"); + dev_err(dev, "sport error: ROVF\n"); return IRQ_HANDLED; } -- 1.7.0.4

4 7

[alsa-devel] [PATCH] ALSA: HDA: Add inverted internal mic quirk for Lenovo S205
by David Henningsson 21 Jun '12

21 Jun '12

The Lenovo Ideapad S205 has an internal mic where the right channel is phase inverted. BugLink: https://bugs.launchpad.net/bugs/884652 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172370b..2af0868 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4466,6 +4466,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; -- 1.7.9.5

2 1

[alsa-devel] why snd_soc_suspend is not called
by zhangfei gao 21 Jun '12

21 Jun '12

Hi, Mark Excuse me, I am confused about snd_soc_suspend. When realizing cpu_dai->driver->suspend, we found snd_soc_suspend is not called. In fact, even soc_probe is not called since no soc-audio device. We manually alloc soc-audio deivce before in platfrom.c. dev = platform_device_alloc("soc-audio", -1); platform_set_drvdata(device, &snd_soc_card); platform_device_add(dev); When running, system recommend using snd_soc_register_card instead. After doing such modification, the audio could work. But strangely snd_soc_suspend is not called, as well as soc-probe. So cpu_dai->driver->suspend will no be called accordingly. If manually add "soc-audio" in arch/, the soc-probe can be called. But snd_soc_suspend still fail to be called. Could you give me some suggestion? Thanks a lot. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 5

[alsa-devel] Tempo event in a Midi sequencer...
by Ru Vuott 21 Jun '12

21 Jun '12

Hello, I am finally able to make a simple midi sequencer with the Gambas language. It works, but I have some little problem about changing the metronomic Tempo within a midi song. Since I used the method of sending an Echo event to send Midi Messages groups to ALSA, I thought I'd solve the problem of an eventual return of metronomic Time by sending an echo and backing from Alsa custom (Type: 90). At the time of receipt of the echo by the Alsa function: int snd_seq_event_input(snd_seq_t *handle, snd_seq_event_t **ev) the code sends the value of the tempo to a routine, where there are these Alsa functions: snd_seq_queue_tempo_malloc(...) snd_seq_queue_tempo_set_tempo(...) snd_seq_set_queue_tempo(...) snd_seq_queue_tempo_free(...) ...obviously with teir parametres. I noticed, though, unfortunately (incomprehensibly) that often the change is a bit 'earlier than required. I do not think there is a problem of Delta Time, since the Delta Time of metronomic Tempo event is treated the same way as that relating to other MIDI events. Looking for a solution, I had a look at the code of your Aplaymidi, and I saw this statement: case 0x51: /* tempo */ ...... ...etc event->type = SND_SEQ_EVENT_TEMPO; ...etc, so, I thought that the change of Tempo occur simply by sending an event by identifying it with precisely the type number: 35. I performed the test, checking the result is listening to the song, seeing both as reported by Aseqdump. I guess it Aseqdump me signals a correct delivery, because it gives me back this response: queue + a number depending on the value sent with the Tempo event. Instead, listening to the midi track, the change does not happen! I am writing to ask you, if my analysis is correct, and if it is sufficient to send only time that event, or if some function of Alsa is absolutely necessary Thanks Vuotttttttt

2 1

Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by zhangfei gao 21 Jun '12

21 Jun '12

On Thu, Jun 14, 2012 at 12:37 AM, Chris Ball <cjb(a)laptop.org> wrote: > Hi Zhangfei, > > On Mon, Jun 11 2012, Zhangfei Gao wrote: >> v1->v2: >> modify accroingly with suggestion from Vinod and Mark >> remove deprecated chan->private and dma.h >> remove prep_slave_sg >> remove module_init and move operation to probe >> misc change >> >> v0->v1: >> dmaengine -> soc-dmaengine. >> update according to Mark's suggestion >> >> These patches provide mmp audio support under alsa via dmaengine >> Support platfrom pxa688 and pxa910 >> mmp_tdma.c and mmp-pcm.c is shared for pxa688 and pxa910 >> pxa688 (mmp2) use mmp2-sspa.c >> pxa910 directly use pxa-ssp.c >> mmp_tdma.c is under dmaengine framework >> >> Verified on brownstone (pxa688) and ttc-dkb > > Great, we're looking into adopting this for OLPC XO-1.75 too. > > Could you explain how system suspend/resume should work with this > patchset? You don't seem to be adding any suspend/resume handlers. > > Thanks, > > - Chris. Chris Some confusion about audio suspend. We use android environment, wake lock is hold preventing system entering suspend when audio work. Only entering early-suspend, which close LCD etc. If this is the case, we may not need consider suspend during audio play back. If support suspend during audio work, some modification required. I found SNDRV_PCM_TRIGGER_START/STOP is send in suspend/resume instead of SNDRV_PCM_TRIGGER_RESUME/SUSPEND as before. The result is dmaengine_pcm_prepare_and_submit will be re-called every suspend/resume. Unfortunately we malloc descriptor in prep function, but no function to free at such case. Thanks -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

Re: [alsa-devel] [PATCH 1/5] dmaengine: mmp_tdma: add mmp tdma support
by Vinod Koul 21 Jun '12

21 Jun '12

On Fri, 2012-06-15 at 11:04 +0800, Zhangfei Gao wrote: > Add support for two-channel dma under dmaengine > support: mmp-adma and pxa910-squ Applied, thanks -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH 1/2 v2] asoc: add sport driver for bf6xx soc
by Scott Jiang 20 Jun '12

20 Jun '12

The SPORT(Serial Port) module on bf6xx soc has a totally different ip comparing to bf5xx soc. An individual SPORT module consists of two independently configurable SPORT halves with identical functionality. Each SPORT half can be configured for either transmitter or receiver. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/Kconfig | 10 +- sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf6xx-sport.c | 422 ++++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf6xx-sport.h | 82 ++++++++ 4 files changed, 513 insertions(+), 3 deletions(-) create mode 100644 sound/soc/blackfin/bf6xx-sport.c create mode 100644 sound/soc/blackfin/bf6xx-sport.h diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 9f6bc55..0374a39 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,7 +1,8 @@ config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI BF5xx chip" + tristate "SoC I2S Audio for the ADI Blackfin chip" depends on BLACKFIN - select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_SPORT if !BF60x + select SND_BF6XX_SOC_SPORT if BF60x help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -162,6 +163,9 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SOC_SPORT tristate +config SND_BF6XX_SOC_SPORT + tristate + config SND_BF5XX_SOC_I2S tristate @@ -173,7 +177,7 @@ config SND_BF5XX_SOC_AC97 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) + depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 1bf86cc..13b0922 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -3,6 +3,7 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o +snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o @@ -11,6 +12,7 @@ obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o +obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c new file mode 100644 index 0000000..f19a72b --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -0,0 +1,422 @@ +/* + * bf6xx_sport.c Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <asm/blackfin.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "bf6xx-sport.h" + +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; + sport->tx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_tx_params); + +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; + sport->rx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_rx_params); + +static int compute_wdsize(size_t wdsize) +{ + switch (wdsize) { + case 1: + return WDSIZE_8 | PSIZE_8; + case 2: + return WDSIZE_16 | PSIZE_16; + default: + return WDSIZE_32 | PSIZE_32; + } +} + +void sport_tx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); + set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->tx_dma_chan); + sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_tx_start); + +void sport_rx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); + set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->rx_dma_chan); + sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_rx_start); + +void sport_tx_stop(struct sport_device *sport) +{ + sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->tx_dma_chan); +} +EXPORT_SYMBOL(sport_tx_stop); + +void sport_rx_stop(struct sport_device *sport) +{ + sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->rx_dma_chan); +} +EXPORT_SYMBOL(sport_rx_stop); + +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data) +{ + sport->tx_callback = tx_callback; + sport->tx_data = tx_data; +} +EXPORT_SYMBOL(sport_set_tx_callback); + +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data) +{ + sport->rx_callback = rx_callback; + sport->rx_data = rx_data; +} +EXPORT_SYMBOL(sport_set_rx_callback); + +static void setup_desc(struct dmasg *desc, void *buf, int fragcount, + size_t fragsize, unsigned int cfg, + unsigned int count, size_t wdsize) +{ + + int i; + + for (i = 0; i < fragcount; ++i) { + desc[i].next_desc_addr = &(desc[i + 1]); + desc[i].start_addr = (unsigned long)buf + i*fragsize; + desc[i].cfg = cfg; + desc[i].x_count = count; + desc[i].x_modify = wdsize; + desc[i].y_count = 0; + desc[i].y_modify = 0; + } + + /* make circular */ + desc[fragcount-1].next_desc_addr = desc; +} + +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + + sport->tx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->tx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->tx_desc) + return -ENOMEM; + + sport->tx_buf = buf; + sport->tx_fragsize = fragsize; + sport->tx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; + + setup_desc(sport->tx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_tx_dma); + +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); + + sport->rx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->rx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->rx_desc) + return -ENOMEM; + + sport->rx_buf = buf; + sport->rx_fragsize = fragsize; + sport->rx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) + | WNR | NDSIZE_6; + + setup_desc(sport->rx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_rx_dma); + +unsigned long sport_curr_offset_tx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); + + return (unsigned char *)curr - sport->tx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_tx); + +unsigned long sport_curr_offset_rx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); + + return (unsigned char *)curr - sport->rx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_rx); + +static irqreturn_t sport_tx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + static unsigned long status; + + status = get_dma_curr_irqstat(sport->tx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->tx_dma_chan); + SSYNC(); + } + if (sport->tx_callback) + sport->tx_callback(sport->tx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_rx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + unsigned long status; + + status = get_dma_curr_irqstat(sport->rx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->rx_dma_chan); + SSYNC(); + } + if (sport->rx_callback) + sport->rx_callback(sport->rx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_err_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + struct device *dev = &sport->pdev->dev; + + if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: TUVF\n"); + if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: ROVF\n"); + + return IRQ_HANDLED; +} + +static int sport_get_resource(struct sport_device *sport) +{ + struct platform_device *pdev = sport->pdev; + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + if (!pdata) { + dev_err(dev, "No platform data\n"); + return -ENODEV; + } + sport->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "No tx MEM resource\n"); + return -ENODEV; + } + sport->tx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + dev_err(dev, "No rx MEM resource\n"); + return -ENODEV; + } + sport->rx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "No tx DMA resource\n"); + return -ENODEV; + } + sport->tx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "No rx DMA resource\n"); + return -ENODEV; + } + sport->rx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "No tx error irq resource\n"); + return -ENODEV; + } + sport->tx_err_irq = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); + if (!res) { + dev_err(dev, "No rx error irq resource\n"); + return -ENODEV; + } + sport->rx_err_irq = res->start; + + return 0; +} + +static int sport_request_resource(struct sport_device *sport) +{ + struct device *dev = &sport->pdev->dev; + int ret; + + ret = peripheral_request_list(sport->pin_req, "soc-audio"); + if (ret) { + dev_err(dev, "Unable to request sport pin\n"); + return ret; + } + + ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); + goto err_tx_dma; + } + set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); + + ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); + goto err_rx_dma; + } + set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); + + ret = request_irq(sport->tx_err_irq, sport_err_irq, + 0, "SPORT TX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); + goto err_tx_irq; + } + + ret = request_irq(sport->rx_err_irq, sport_err_irq, + 0, "SPORT RX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); + goto err_rx_irq; + } + + return 0; +err_rx_irq: + free_irq(sport->tx_err_irq, sport); +err_tx_irq: + free_dma(sport->rx_dma_chan); +err_rx_dma: + free_dma(sport->tx_dma_chan); +err_tx_dma: + peripheral_free_list(sport->pin_req); + return ret; +} + +static void sport_free_resource(struct sport_device *sport) +{ + free_irq(sport->rx_err_irq, sport); + free_irq(sport->tx_err_irq, sport); + free_dma(sport->rx_dma_chan); + free_dma(sport->tx_dma_chan); + peripheral_free_list(sport->pin_req); +} + +struct sport_device *sport_create(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct sport_device *sport; + int ret; + + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "Unable to allocate memory for sport device\n"); + return NULL; + } + sport->pdev = pdev; + + ret = sport_get_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + ret = sport_request_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + dev_dbg(dev, "SPORT create success\n"); + return sport; +} +EXPORT_SYMBOL(sport_create); + +void sport_delete(struct sport_device *sport) +{ + sport_free_resource(sport); +} +EXPORT_SYMBOL(sport_delete); + +MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h new file mode 100644 index 0000000..307d193 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.h @@ -0,0 +1,82 @@ +/* + * bf6xx_sport - Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef _BF6XX_SPORT_H_ +#define _BF6XX_SPORT_H_ + +#include <linux/platform_device.h> +#include <asm/bfin_sport3.h> + +struct sport_device { + struct platform_device *pdev; + const unsigned short *pin_req; + struct sport_register *tx_regs; + struct sport_register *rx_regs; + int tx_dma_chan; + int rx_dma_chan; + int tx_err_irq; + int rx_err_irq; + + void (*tx_callback)(void *data); + void *tx_data; + void (*rx_callback)(void *data); + void *rx_data; + + struct dmasg *tx_desc; + struct dmasg *rx_desc; + unsigned int tx_desc_size; + unsigned int rx_desc_size; + unsigned char *tx_buf; + unsigned char *rx_buf; + unsigned int tx_fragsize; + unsigned int rx_fragsize; + unsigned int tx_frags; + unsigned int rx_frags; + unsigned int wdsize; +}; + +struct sport_params { + u32 spctl; + u32 div; +}; + +struct sport_device *sport_create(struct platform_device *pdev); +void sport_delete(struct sport_device *sport); +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params); +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params); +void sport_tx_start(struct sport_device *sport); +void sport_rx_start(struct sport_device *sport); +void sport_tx_stop(struct sport_device *sport); +void sport_rx_stop(struct sport_device *sport); +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data); +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data); +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +unsigned long sport_curr_offset_tx(struct sport_device *sport); +unsigned long sport_curr_offset_rx(struct sport_device *sport); + + + +#endif -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ALSA: Add missing include of pcm.h to pcm_params.h
by Mark Brown 20 Jun '12

20 Jun '12

There's a dependency but no #include. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm_params.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index f494f1e..37ae12e 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -22,6 +22,8 @@ * */ +#include <sound/pcm.h> + int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir); -- 1.7.10

2 3

[alsa-devel] [PATCH v2 1/3] dmaengine: Add wrapper for device_tx_status callback
by Lars-Peter Clausen 20 Jun '12

20 Jun '12

This patch adds a small inline wrapper for the devivce_tx_status callback of a dma device. This makes the source code of users of this function a bit more compact and a bit more legible. E.g.: -status = chan->device->device_tx_status(chan, cookie, &state) +status = dmaengine_tx_status(chan, cookie, &state) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1 --- include/linux/dmaengine.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 56377df..cc0756a 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -670,6 +670,12 @@ static inline int dmaengine_resume(struct dma_chan *chan) return dmaengine_device_control(chan, DMA_RESUME, 0); } +static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *state) +{ + return chan->device->device_tx_status(chan, cookie, state); +} + static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc) { return desc->tx_submit(desc); -- 1.7.10

4 13

[alsa-devel] [PATCH] ASoC: wm2200: Add missing BCLK rate
by Mark Brown 20 Jun '12

20 Jun '12

Without this very high BCLKs will be configured incorrectly. Reported-by: Axel Lin <axel.lin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index acbdc5f..32682c1 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1491,6 +1491,7 @@ static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = { static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = { 5644800, + 3763200, 2882400, 1881600, 1411200, -- 1.7.10

1 0

[alsa-devel] [PATCH 1/2] swarm_cs4297: Provide definitions for AC97 registers
by Ezequiel Garcia 20 Jun '12

20 Jun '12

This patch removes the last usage of linux/ac97_codec.h by re-defining used AC97 registers. We can't use sound/ac97_codec.h here, since it is an OSS driver. Not even compilation tested. Cc: Ralf Baechle <ralf(a)linux-mips.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Ezequiel Garcia <elezegarcia(a)gmail.com> --- sound/oss/swarm_cs4297a.c | 17 ++++++++++++++++- 1 files changed, 16 insertions(+), 1 deletions(-) diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 09d4648..bd75f54 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -69,7 +69,6 @@ #include <linux/sound.h> #include <linux/slab.h> #include <linux/soundcard.h> -#include <linux/ac97_codec.h> #include <linux/pci.h> #include <linux/bitops.h> #include <linux/interrupt.h> @@ -199,6 +198,22 @@ static const char invalid_magic[] = } \ }) +/* AC97 registers */ +#define AC97_MASTER_VOL_STEREO 0x0002 /* Line Out */ +#define AC97_PCBEEP_VOL 0x000a /* none */ +#define AC97_PHONE_VOL 0x000c /* TAD Input (mono) */ +#define AC97_MIC_VOL 0x000e /* MIC Input (mono) */ +#define AC97_LINEIN_VOL 0x0010 /* Line Input (stereo) */ +#define AC97_CD_VOL 0x0012 /* CD Input (stereo) */ +#define AC97_AUX_VOL 0x0016 /* Aux Input (stereo) */ +#define AC97_PCMOUT_VOL 0x0018 /* Wave Output (stereo) */ +#define AC97_RECORD_SELECT 0x001a /* */ +#define AC97_RECORD_GAIN 0x001c +#define AC97_GENERAL_PURPOSE 0x0020 +#define AC97_3D_CONTROL 0x0022 +#define AC97_POWER_CONTROL 0x0026 +#define AC97_VENDOR_ID1 0x007c + struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs }; typedef struct serdma_descr_s { -- 1.7.4.4

2 2

Re: [alsa-devel] [PATCH 1/1] ASoC: lm49453: Fix compilation warning
by Reddy, M.R. Swami 20 Jun '12

20 Jun '12

This compilation warning issue was fixed. Patch submitted by Mark. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052425.html Please use the latest source repo. Thanks Swami -----Original Message----- From: Sachin Kamat [mailto:sachin.kamat@linaro.org] Sent: Wednesday, June 20, 2012 11:38 AM To: alsa-devel(a)alsa-project.org Cc: broonie(a)opensource.wolfsonmicro.com; Reddy, M.R. Swami; sachin.kamat(a)linaro.org; patches(a)linaro.org Subject: [PATCH 1/1] ASoC: lm49453: Fix compilation warning Fixes the following build warning: sound/soc/codecs/lm49453.c: In function ‘lm49453_i2c_probe’: sound/soc/codecs/lm49453.c:1510:24: warning: passing argument 3 of ‘snd_soc_register_codec’ discards ‘const’ qualifier from pointer target type [enabled by default] include/sound/soc.h:333:5: note: expected ‘struct snd_soc_dai_driver *’ but argument is of type ‘const struct snd_soc_dai_driver *’ The struct snd_soc_dai_driver lm49453_dai[] has been made non-const, in line with other snd_soc_dai_driver structure variables in other codec files. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/lm49453.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index c1bc945..99b0a9d 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1357,7 +1357,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { -- 1.7.4.1

1 0

Re: [alsa-devel] [PATCH 0/2] snd_usb_audio: ignore ctrl errors on *
by Takashi Iwai 20 Jun '12

20 Jun '12

At Tue, 19 Jun 2012 22:21:17 +0200, Oleksij Rempel wrote: > > This two patches fix stability issues for two logitech cams. > I assume there is more of them, there are some bug reports on > different trackers which describe same symptoms. > > In my case i was able to reproduce this bug mostly with skype > capturing video and audio over usb on webcam. > E3500 can't work at all with pulseaudio without this patch. > The Notebook Pro cam can work but some times it get unstable. > > Ignoring ctrl errors seems to fix both cases. Thanks. Maybe digging deeper to look for the real culprit would be better, but your bandaid looks simple enough, so I applied the patches now. Takashi

1 0

[alsa-devel] alsamixer - Mute AC97 "PCM Playback Switch" also mute "Master Playback Switch" and "Surround Playback Switch"
by Raymond Yau 20 Jun '12

20 Jun '12

Tested on Ubuntu 12.04 and Fedora 17 Live CD When PA server is running , using alsamixer -c0 to mute AC97 's "PCM Playback Switch" also mute "Master Playback Switch" and "Surround Playback Switch" but unmute "PCM Playback Switch" does not unmute "Master Playback Switch" and "Surround Playback Switch" of stac9708 alsamixer behave normally when shutdown PA server by echo "autospwan=off" > .pulse/client.conf and pulseaudio -k Why pulseaudio server behave in this way ?

1 0

[alsa-devel] [PATCH 1/3] maestro3: Remove unused AC97 register definitions
by Ezequiel Garcia 19 Jun '12

19 Jun '12

Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Ezequiel Garcia <elezegarcia(a)gmail.com> --- sound/pci/maestro3.c | 68 -------------------------------------------------- 1 files changed, 0 insertions(+), 68 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index deef213..adb3b4c 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -361,74 +361,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define DSP2HOST_REQ_I2SRATE 0x02 #define DSP2HOST_REQ_TIMER 0x04 -/* AC97 registers */ -/* XXX fix this crap up */ -/*#define AC97_RESET 0x00*/ - -#define AC97_VOL_MUTE_B 0x8000 -#define AC97_VOL_M 0x1F -#define AC97_LEFT_VOL_S 8 - -#define AC97_MASTER_VOL 0x02 -#define AC97_LINE_LEVEL_VOL 0x04 -#define AC97_MASTER_MONO_VOL 0x06 -#define AC97_PC_BEEP_VOL 0x0A -#define AC97_PC_BEEP_VOL_M 0x0F -#define AC97_SROUND_MASTER_VOL 0x38 -#define AC97_PC_BEEP_VOL_S 1 - -/*#define AC97_PHONE_VOL 0x0C -#define AC97_MIC_VOL 0x0E*/ -#define AC97_MIC_20DB_ENABLE 0x40 - -/*#define AC97_LINEIN_VOL 0x10 -#define AC97_CD_VOL 0x12 -#define AC97_VIDEO_VOL 0x14 -#define AC97_AUX_VOL 0x16*/ -#define AC97_PCM_OUT_VOL 0x18 -/*#define AC97_RECORD_SELECT 0x1A*/ -#define AC97_RECORD_MIC 0x00 -#define AC97_RECORD_CD 0x01 -#define AC97_RECORD_VIDEO 0x02 -#define AC97_RECORD_AUX 0x03 -#define AC97_RECORD_MONO_MUX 0x02 -#define AC97_RECORD_DIGITAL 0x03 -#define AC97_RECORD_LINE 0x04 -#define AC97_RECORD_STEREO 0x05 -#define AC97_RECORD_MONO 0x06 -#define AC97_RECORD_PHONE 0x07 - -/*#define AC97_RECORD_GAIN 0x1C*/ -#define AC97_RECORD_VOL_M 0x0F - -/*#define AC97_GENERAL_PURPOSE 0x20*/ -#define AC97_POWER_DOWN_CTRL 0x26 -#define AC97_ADC_READY 0x0001 -#define AC97_DAC_READY 0x0002 -#define AC97_ANALOG_READY 0x0004 -#define AC97_VREF_ON 0x0008 -#define AC97_PR0 0x0100 -#define AC97_PR1 0x0200 -#define AC97_PR2 0x0400 -#define AC97_PR3 0x0800 -#define AC97_PR4 0x1000 - -#define AC97_RESERVED1 0x28 - -#define AC97_VENDOR_TEST 0x5A - -#define AC97_CLOCK_DELAY 0x5C -#define AC97_LINEOUT_MUX_SEL 0x0001 -#define AC97_MONO_MUX_SEL 0x0002 -#define AC97_CLOCK_DELAY_SEL 0x1F -#define AC97_DAC_CDS_SHIFT 6 -#define AC97_ADC_CDS_SHIFT 11 - -#define AC97_MULTI_CHANNEL_SEL 0x74 - -/*#define AC97_VENDOR_ID1 0x7C -#define AC97_VENDOR_ID2 0x7E*/ - /* * ASSP control regs */ -- 1.7.4.4

2 9

[alsa-devel] [PATCH] ASoC: wm8904: Remove unused variable in probe()
by Mark Brown 19 Jun '12

19 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index cc5fb43..0013afe 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2054,7 +2054,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - struct wm8904_pdata *pdata = wm8904->pdata; int ret; codec->control_data = wm8904->regmap; -- 1.7.10

1 0

Re: [alsa-devel] [PATCH v2 1/5] dmaengine: mmp_tdma: add mmp tdma support
by Vinod Koul 19 Jun '12

19 Jun '12

On Mon, 2012-06-11 at 18:04 +0800, Zhangfei Gao wrote: > +static int mmp_tdma_clear_chan_irq(struct mmp_tdma_chan *tdmac) > +{ > + u32 reg = readl(tdmac->reg_base + TDISR); > + > + if (reg & TDISR_COMP) { > + /* clear irq */ > + reg &= ~TDISR_COMP; > + writel(reg, tdmac->reg_base + TDISR); > + > + return 0; > + } > + return -1; > +} -1? Please return a proper linux error code. -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ASoC: wm5100: Remove stubs of ASoC-level register map code
by Mark Brown 19 Jun '12

19 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ba52534..33aa363 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2517,13 +2517,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } -static int wm5100_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - - static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, @@ -2531,8 +2524,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, .idle_bias_off = 1, - .reg_cache_size = WM5100_MAX_REGISTER, - .volatile_register = wm5100_soc_volatile, .seq_notifier = wm5100_seq_notifier, .controls = wm5100_snd_controls, -- 1.7.10

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[alsa-devel] [PATCH V2 13/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 19 Jun '12

19 Jun '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch make it usable: 1, Use 32-bit DMA as a workaround (Loongson has a hardware bug that it doesn't support DMA address above 4GB). 2, Modify patch_conexant.c to add Lemote specific code. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 + sound/pci/hda/hda_intel.c | 5 ++++ sound/pci/hda/patch_conexant.c | 52 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 57 insertions(+), 2 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ab741b0..d8b0a52 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2906,3 +2906,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6392b..2b73ed4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3013,6 +3013,11 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) gcap &= ~ICH6_GCAP_64OK; } +#ifdef CONFIG_CPU_LOONGSON3 + /* Workaround: Loongson 3 doesn't support 64-bit DMA */ + gcap &= ~ICH6_GCAP_64OK; +#endif + /* disable buffer size rounding to 128-byte multiples if supported */ if (align_buffer_size >= 0) chip->align_buffer_size = !!align_buffer_size; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3acb582..1c8dfb9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int lemote:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; @@ -2278,7 +2279,7 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); - else if (spec->asus) + else if (spec->asus || spec->lemote) cxt5066_asus_automic(codec); } @@ -2911,6 +2912,32 @@ static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_lemote[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1: Connection index: 0 */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* ADC2, ADC3: unused */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2948,6 +2975,8 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_LEMOTE_A1004, /* Lemote Laptop A1004 */ + CXT5066_LEMOTE_A1205, /* Lemote All-In-One A1205 */ CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -2961,6 +2990,8 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_LEMOTE_A1004] = "lemote-laptop-a1004", + [CXT5066_LEMOTE_A1205] = "lemote-aio-a1205", [CXT5066_AUTO] = "auto", }; @@ -2993,6 +3024,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT5066_LEMOTE_A1004), + SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT5066_LEMOTE_A1205), SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3073,7 +3106,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->mic_boost = 3; /* default 30dB gain */ break; - + case CXT5066_LEMOTE_A1004: + case CXT5066_LEMOTE_A1205: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_lemote; + spec->num_init_verbs++; + spec->lemote = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.7.3

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Re: [alsa-devel] ASoC: codecs: Add AB8500 codec-driver
by Dan Carpenter 19 Jun '12

19 Jun '12

Hello Ola Lilja, The patch 679d7abdc754: "ASoC: codecs: Add AB8500 codec-driver" from Jun 7, 2012, leads to the following Sparse warnings: sound/soc/codecs/ab8500-codec.c:1960:53: warning: cast truncates bits from constant value (1013 becomes 13) sound/soc/codecs/ab8500-codec.c:1968:53: warning: cast truncates bits from constant value (1013 becomes 13) sound/soc/codecs/ab8500-codec.c 1958 /* Set DMic-clocks to outputs */ 1959 status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, 1960 (u8)AB8500_GPIO_DIR4_REG, ^^^^^^^^^^^^^^^^^^^^ 1961 &value8); 1962 if (status < 0) 1963 return status; 1964 value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | 1965 GPIO31_DIR_OUTPUT; 1966 status = abx500_set_register_interruptible(codec->dev, 1967 (u8)AB8500_MISC, 1968 (u8)AB8500_GPIO_DIR4_REG, ^^^^^^^^^^^^^^^^^^^^^^^^ 1969 value); In my quick check we only seem to use the truncated values of these defines so I'm not sure this was intended. regards, dan carpenter

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Re: [alsa-devel] [PATCH 3/3] ASoC: follow the updated samsung DMA common operations
by Kukjin Kim 19 Jun '12

19 Jun '12

Boojin Kim wrote: > > Mark Brown wrote > > > This patch uses config() function to configure DMA transmit options. > > > > Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > Thanks for your ack and let me pick this up in Samsung tree. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd. > > It'd be good if we could get all the older Samsung platforms moved over > to dmaengine... > Thanks for your ack :) > And, Our engineer's preparing to move the older Samsung dma driver into > dmaengine.

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Re: [alsa-devel] [PATCH 1/3] ARM: SAMSUNG: Add config() function in DMA common operations
by Kukjin Kim 19 Jun '12

19 Jun '12

Boojin Kim wrote: > > This patch adds config() that configures DMA transmit option. > This function was originally included in request(). > But, Some DMA client driver requires to change the configuration after > request(). > So, This patch picks up it from request(). > > Signed-off-by: Boojin Kim <boojin.kim(a)samsung.com> > --- > arch/arm/plat-samsung/dma-ops.c | 77 ++++++++++++++----------- > arch/arm/plat-samsung/include/plat/dma-ops.h | 20 ++++--- > arch/arm/plat-samsung/s3c-dma-ops.c | 39 +++++++------ > 3 files changed, 77 insertions(+), 59 deletions(-) > > diff --git a/arch/arm/plat-samsung/dma-ops.c b/arch/arm/plat-samsung/dma- > ops.c > index eb9f4f5..51d8590 100644 > --- a/arch/arm/plat-samsung/dma-ops.c > +++ b/arch/arm/plat-samsung/dma-ops.c > @@ -19,72 +19,80 @@ > #include <mach/dma.h> > > static unsigned samsung_dmadev_request(enum dma_ch dma_ch, > - struct samsung_dma_info *info) > + struct samsung_dma_req *param) > { > - struct dma_chan *chan; > dma_cap_mask_t mask; > - struct dma_slave_config slave_config; > void *filter_param; > > dma_cap_zero(mask); > - dma_cap_set(info->cap, mask); > + dma_cap_set(param->cap, mask); > > /* > * If a dma channel property of a device node from device tree is > * specified, use that as the fliter parameter. > */ > - filter_param = (dma_ch == DMACH_DT_PROP) ? (void *)info- > >dt_dmach_prop : > - (void *)dma_ch; > - chan = dma_request_channel(mask, pl330_filter, filter_param); > + filter_param = (dma_ch == DMACH_DT_PROP) ? > + (void *)param->dt_dmach_prop : (void *)dma_ch; > + return (unsigned)dma_request_channel(mask, pl330_filter, > filter_param); > +} > > - if (info->direction == DMA_DEV_TO_MEM) { > +static int samsung_dmadev_release(unsigned ch, > + struct s3c2410_dma_client *client) > +{ > + dma_release_channel((struct dma_chan *)ch); > + > + return 0; > +} > + > +static int samsung_dmadev_config(unsigned ch, > + struct samsung_dma_config *param) > +{ > + struct dma_chan *chan = (struct dma_chan *)ch; > + struct dma_slave_config slave_config; > + > + if (param->direction == DMA_DEV_TO_MEM) { > memset(&slave_config, 0, sizeof(struct dma_slave_config)); > - slave_config.direction = info->direction; > - slave_config.src_addr = info->fifo; > - slave_config.src_addr_width = info->width; > + slave_config.direction = param->direction; > + slave_config.src_addr = param->fifo; > + slave_config.src_addr_width = param->width; > slave_config.src_maxburst = 1; > dmaengine_slave_config(chan, &slave_config); > - } else if (info->direction == DMA_MEM_TO_DEV) { > + } else if (param->direction == DMA_MEM_TO_DEV) { > memset(&slave_config, 0, sizeof(struct dma_slave_config)); > - slave_config.direction = info->direction; > - slave_config.dst_addr = info->fifo; > - slave_config.dst_addr_width = info->width; > + slave_config.direction = param->direction; > + slave_config.dst_addr = param->fifo; > + slave_config.dst_addr_width = param->width; > slave_config.dst_maxburst = 1; > dmaengine_slave_config(chan, &slave_config); > + } else { > + printk(KERN_WARNING "unsupported direction\n"); > + return -EINVAL; > } > > - return (unsigned)chan; > -} > - > -static int samsung_dmadev_release(unsigned ch, > - struct s3c2410_dma_client *client) > -{ > - dma_release_channel((struct dma_chan *)ch); > - > return 0; > } > > static int samsung_dmadev_prepare(unsigned ch, > - struct samsung_dma_prep_info *info) > + struct samsung_dma_prep *param) > { > struct scatterlist sg; > struct dma_chan *chan = (struct dma_chan *)ch; > struct dma_async_tx_descriptor *desc; > > - switch (info->cap) { > + switch (param->cap) { > case DMA_SLAVE: > sg_init_table(&sg, 1); > - sg_dma_len(&sg) = info->len; > - sg_set_page(&sg, pfn_to_page(PFN_DOWN(info->buf)), > - info->len, offset_in_page(info->buf)); > - sg_dma_address(&sg) = info->buf; > + sg_dma_len(&sg) = param->len; > + sg_set_page(&sg, pfn_to_page(PFN_DOWN(param->buf)), > + param->len, offset_in_page(param->buf)); > + sg_dma_address(&sg) = param->buf; > > desc = dmaengine_prep_slave_sg(chan, > - &sg, 1, info->direction, DMA_PREP_INTERRUPT); > + &sg, 1, param->direction, DMA_PREP_INTERRUPT); > break; > case DMA_CYCLIC: > - desc = dmaengine_prep_dma_cyclic(chan, > - info->buf, info->len, info->period, info->direction); > + desc = dmaengine_prep_dma_cyclic(chan, param->buf, > + param->len, param->period, param->direction); > break; > default: > dev_err(&chan->dev->device, "unsupported format\n"); > @@ -96,8 +104,8 @@ static int samsung_dmadev_prepare(unsigned ch, > return -EFAULT; > } > > - desc->callback = info->fp; > - desc->callback_param = info->fp_param; > + desc->callback = param->fp; > + desc->callback_param = param->fp_param; > > dmaengine_submit((struct dma_async_tx_descriptor *)desc); > > @@ -119,6 +127,7 @@ static inline int samsung_dmadev_flush(unsigned ch) > static struct samsung_dma_ops dmadev_ops = { > .request = samsung_dmadev_request, > .release = samsung_dmadev_release, > + .config = samsung_dmadev_config, > .prepare = samsung_dmadev_prepare, > .trigger = samsung_dmadev_trigger, > .started = NULL, > diff --git a/arch/arm/plat-samsung/include/plat/dma-ops.h b/arch/arm/plat- > samsung/include/plat/dma-ops.h > index 71a6827..f5144cd 100644 > --- a/arch/arm/plat-samsung/include/plat/dma-ops.h > +++ b/arch/arm/plat-samsung/include/plat/dma-ops.h > @@ -16,7 +16,13 @@ > #include <linux/dmaengine.h> > #include <mach/dma.h> > > -struct samsung_dma_prep_info { > +struct samsung_dma_req { > + enum dma_transaction_type cap; > + struct property *dt_dmach_prop; > + struct s3c2410_dma_client *client; > +}; > + > +struct samsung_dma_prep { > enum dma_transaction_type cap; > enum dma_transfer_direction direction; > dma_addr_t buf; > @@ -26,19 +32,17 @@ struct samsung_dma_prep_info { > void *fp_param; > }; > > -struct samsung_dma_info { > - enum dma_transaction_type cap; > +struct samsung_dma_config { > enum dma_transfer_direction direction; > enum dma_slave_buswidth width; > dma_addr_t fifo; > - struct s3c2410_dma_client *client; > - struct property *dt_dmach_prop; > }; > > struct samsung_dma_ops { > - unsigned (*request)(enum dma_ch ch, struct samsung_dma_info *info); > - int (*release)(unsigned ch, struct s3c2410_dma_client *client); > - int (*prepare)(unsigned ch, struct samsung_dma_prep_info *info); > + unsigned (*request)(enum dma_ch ch, struct samsung_dma_req *param); > + int (*release)(unsigned ch, void *param); > + int (*config)(unsigned ch, struct samsung_dma_config *param); > + int (*prepare)(unsigned ch, struct samsung_dma_prep *param); > int (*trigger)(unsigned ch); > int (*started)(unsigned ch); > int (*flush)(unsigned ch); > diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat- > samsung/s3c-dma-ops.c > index 7814949..f99448c 100644 > --- a/arch/arm/plat-samsung/s3c-dma-ops.c > +++ b/arch/arm/plat-samsung/s3c-dma-ops.c > @@ -36,30 +36,26 @@ static void s3c_dma_cb(struct s3c2410_dma_chan > *channel, void *param, > } > > static unsigned s3c_dma_request(enum dma_ch dma_ch, > - struct samsung_dma_info *info) > + struct samsung_dma_req *param) > { > struct cb_data *data; > > - if (s3c2410_dma_request(dma_ch, info->client, NULL) < 0) { > - s3c2410_dma_free(dma_ch, info->client); > + if (s3c2410_dma_request(dma_ch, param->client, NULL) < 0) { > + s3c2410_dma_free(dma_ch, param->client); > return 0; > } > > + if (param->cap == DMA_CYCLIC) > + s3c2410_dma_setflags(dma_ch, S3C2410_DMAF_CIRCULAR); > + > data = kzalloc(sizeof(struct cb_data), GFP_KERNEL); > data->ch = dma_ch; > list_add_tail(&data->node, &dma_list); > > - s3c2410_dma_devconfig(dma_ch, info->direction, info->fifo); > - > - if (info->cap == DMA_CYCLIC) > - s3c2410_dma_setflags(dma_ch, S3C2410_DMAF_CIRCULAR); > - > - s3c2410_dma_config(dma_ch, info->width); > - > return (unsigned)dma_ch; > } > > -static int s3c_dma_release(unsigned ch, struct s3c2410_dma_client *client) > +static int s3c_dma_release(unsigned ch, void *param) > { > struct cb_data *data; > > @@ -68,16 +64,24 @@ static int s3c_dma_release(unsigned ch, struct > s3c2410_dma_client *client) > break; > list_del(&data->node); > > - s3c2410_dma_free(ch, client); > + s3c2410_dma_free(ch, param); > kfree(data); > > return 0; > } > > -static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep_info > *info) > +static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param) > +{ > + s3c2410_dma_devconfig(ch, param->direction, param->fifo); > + s3c2410_dma_config(ch, param->width); > + > + return 0; > +} > + > +static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param) > { > struct cb_data *data; > - int len = (info->cap == DMA_CYCLIC) ? info->period : info->len; > + int len = (param->cap == DMA_CYCLIC) ? param->period : param->len; > > list_for_each_entry(data, &dma_list, node) > if (data->ch == ch) > @@ -85,11 +89,11 @@ static int s3c_dma_prepare(unsigned ch, struct > samsung_dma_prep_info *info) > > if (!data->fp) { > s3c2410_dma_set_buffdone_fn(ch, s3c_dma_cb); > - data->fp = info->fp; > - data->fp_param = info->fp_param; > + data->fp = param->fp; > + data->fp_param = param->fp_param; > } > > - s3c2410_dma_enqueue(ch, (void *)data, info->buf, len); > + s3c2410_dma_enqueue(ch, (void *)data, param->buf, len); > > return 0; > } > @@ -117,6 +121,7 @@ static inline int s3c_dma_stop(unsigned ch) > static struct samsung_dma_ops s3c_dma_ops = { > .request = s3c_dma_request, > .release = s3c_dma_release, > + .config = s3c_dma_config, > .prepare = s3c_dma_prepare, > .trigger = s3c_dma_trigger, > .started = s3c_dma_started, > -- > 1.7.1 Looks good to me but need to ack on spi patch in this series. I think, this series should be picked up together. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

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Re: [alsa-devel] [PATCH 2/3] spi: Add the use of DMA config operation
by Kukjin Kim 19 Jun '12

19 Jun '12

Boojin Kim wrote: > > Config operation is separated from request operation in DMA common > operation. > Because spi driver can change the DMA config for every transfer. > So this patch is using the separated DMA config operation. > > Signed-off-by: Boojin Kim <boojin.kim(a)samsung.com> > Signed-off-by: Kyoungil Kim <ki0351.kim(a)samsung.com> > --- > drivers/spi/spi-s3c64xx.c | 33 +++++++++++++++++++-------------- > 1 files changed, 19 insertions(+), 14 deletions(-) > > diff --git a/drivers/spi/spi-s3c64xx.c b/drivers/spi/spi-s3c64xx.c > index 15f864d..f4e2341 100644 > --- a/drivers/spi/spi-s3c64xx.c > +++ b/drivers/spi/spi-s3c64xx.c > @@ -262,14 +262,24 @@ static void prepare_dma(struct s3c64xx_spi_dma_data > *dma, > unsigned len, dma_addr_t buf) > { > struct s3c64xx_spi_driver_data *sdd; > - struct samsung_dma_prep_info info; > + struct samsung_dma_prep info; > + struct samsung_dma_config config; > > - if (dma->direction == DMA_DEV_TO_MEM) > + if (dma->direction == DMA_DEV_TO_MEM) { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, rx_dma); > - else > + config.direction = sdd->rx_dma.direction; > + config.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > + config.width = sdd->cur_bpw / 8; > + sdd->ops->config(sdd->rx_dma.ch, &config); > + } else { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, tx_dma); > + config.direction = sdd->tx_dma.direction; > + config.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > + config.width = sdd->cur_bpw / 8; > + sdd->ops->config(sdd->tx_dma.ch, &config); > + } > > info.cap = DMA_SLAVE; > info.len = len; > @@ -284,20 +294,15 @@ static void prepare_dma(struct s3c64xx_spi_dma_data > *dma, > > static int acquire_dma(struct s3c64xx_spi_driver_data *sdd) > { > - struct samsung_dma_info info; > + struct samsung_dma_req req; > > sdd->ops = samsung_dma_get_ops(); > > - info.cap = DMA_SLAVE; > - info.client = &s3c64xx_spi_dma_client; > - info.width = sdd->cur_bpw / 8; > - > - info.direction = sdd->rx_dma.direction; > - info.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > - sdd->rx_dma.ch = sdd->ops->request(sdd->rx_dma.dmach, &info); > - info.direction = sdd->tx_dma.direction; > - info.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > - sdd->tx_dma.ch = sdd->ops->request(sdd->tx_dma.dmach, &info); > + req.cap = DMA_SLAVE; > + req.client = &s3c64xx_spi_dma_client; > + > + sdd->rx_dma.ch = sdd->ops->request(sdd->rx_dma.dmach, &req); > + sdd->tx_dma.ch = sdd->ops->request(sdd->tx_dma.dmach, &req); > > return 1; > } > -- > 1.7.1 Hi Grant, I think, this is needed to support Samsung SPI with DMA and this has dependency with other patches in this series. If you're ok on this, please reply with your ack :) so that I can take this series together. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

1 0

[alsa-devel] [PATCH] ALSA-LIB - remove Logitech Web Camera from blacklist
by Raymond Yau 19 Jun '12

19 Jun '12

- the commit which add Logitech Web Camera to the blacklist is redundant since Web Camera does not has any playback device and Pulseaudio does not support digital capture

2 2

Re: [alsa-devel] [PATCH 2/2] sound: tegra_alc5632: Adjust to of_get_named_gpio() change
by Leon Romanovsky 19 Jun '12

19 Jun '12

On Mon, Jun 18, 2012 at 8:28 PM, Alexandre Pereira da Silva <aletes.xgr(a)gmail.com> wrote: > On Mon, Jun 18, 2012 at 1:42 PM, Roland Stigge <stigge(a)antcom.de> wrote: >> of_get_named_gpio() was changed to return -EPROBE_DEFER in case of >> gpios not probed yet. This patch adjusts tegra_alc5632 to this. >> >> Signed-off-by: Roland Stigge <stigge(a)antcom.de> > Acked-by: Alexandre Pereira da Silva <aletes.xgr(a)gmail.com> Hi Alexandre, Sorry to be rude, but can you explain me why did you "Acked-by" on this patch ? I don't see your name in git log for tegra_alc5632.c file: http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git;a=history;f=sou… Thanks. -- Leon Romanovsky | Independent Linux Consultant www.leon.nu | leon(a)leon.nu

1 0

Re: [alsa-devel] [PATCH] sound/gpio: of_get_named_gpio() return -EPROBE_DEFER if GPIO deferred
by Stephen Warren 18 Jun '12

18 Jun '12

On 06/18/2012 11:22 AM, Roland Stigge wrote: > of_get_named_gpio_flags() and of_get_named_gpio() return -EPROBE_DEFER if the > respective GPIO is not (yet) available. This is useful if driver's probe() > functions try to get a GPIO whose controller isn't probed yet. Thus, the driver > can be probed again later on. > > The function still returns -EINVAL on other errors (parse error or node doesn't > exist). This way, the case of an optional/intentionally missing GPIO is handled > appropriately. > > For this change, two sound drivers (tegra_{alc5632,wm8903}) need to be > adjusted. Acked-by: Stephen Warren <swarren(a)wwwdotorg.org> (That applies to the two separate sound patches too, if needed!)

1 0

Re: [alsa-devel] [PATCH 1/2] sound: tegra_wm8903: Adjust to of_get_named_gpio() change
by Mark Brown 18 Jun '12

18 Jun '12

On Mon, Jun 18, 2012 at 07:00:14PM +0200, Roland Stigge wrote: > Basically, yes. But this would mean that doing via gpio, we would need > to pull other stuff from the sound tree over to gpio to build upon > because "my" tegra_* sound patches won't apply to the gpio branch currently. Oh, yes. Stephen's gone and reorganised all that code which means it's not a trivial merge. Ho hum. I'm happy to apply them if folks are comfortable with it, I guess ideally I'd be able to pull a branch with the gpio changes in and then apply on top of that.

1 0

Re: [alsa-devel] [PATCH 1/2] sound: tegra_wm8903: Adjust to of_get_named_gpio() change
by Mark Brown 18 Jun '12

18 Jun '12

On Mon, Jun 18, 2012 at 06:42:21PM +0200, Roland Stigge wrote: > of_get_named_gpio() was changed to return -EPROBE_DEFER in case of > gpios not probed yet. This patch adjusts tegra_wm8903 to this. So, I saw from the other thread that this didn't actually happen yet - I guess the best thing is to merge this along with the changes for introducing -EPROBE_DEFER. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] How to redefine "default" to be equivalent to "sysdefault"?
by Tanu Kaskinen 18 Jun '12

18 Jun '12

Hi, I have a file with the following contents: # This file an alsa configuration file, used by pasuspender to make the # "default" Alsa device work while PulseAudio is in suspended state. # Include the system configuration first. Is there any easier way to do this? # 14 lines for including one file is a bit much... @hooks [ { func load files [ { @func concat strings [ { @func datadir } "/alsa.conf" ] } ] } ] # Redefine "default" to be the same as "sysdefault". On most systems # "sysdefault" is configured to use dmix. pcm.!default sysdefault That doesn't do what I want it to do. When I run ALSA_CONFIG_PATH=/path/to/file.conf aplay -Ddefault ~/test.wav aplay still uses pulseaudio. Using "-Dsysdefault" instead of "-Ddefault" doesn't use pulseaudio, so "sysdefault" does what it's supposed to do. The problem is then: why doesn't "default" behave the same way as "sysdefault"? What error did I make? Below is "aplay -L" output with and without setting the environment variable. When the environment variable is set, "default" is listed twice, and "null" and "pulse" are included in the list while they are not included in the list when running aplay without the environment variable. What causes those differences? > ALSA_CONFIG_PATH=/path/to/file.conf aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) pulse Playback/recording through the PulseAudio sound server default Playback/recording through the PulseAudio sound server sysdefault:CARD=Intel HDA Intel, AD198x Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, AD198x Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, AD198x Digital IEC958 (S/PDIF) Digital Audio Output > aplay -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=Intel HDA Intel, AD198x Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, AD198x Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, AD198x Digital IEC958 (S/PDIF) Digital Audio Output -- Tanu

2 4

[alsa-devel] [PATCH] ALSA: Add snd_pcm_rate_bit_to_rate()
by Dimitris Papastamos 18 Jun '12

18 Jun '12

This is essentially the reverse of snd_pcm_rate_to_rate_bit(). This is generally useful as the Compress API uses the rate bit directly and it helps to be able to map back to the actual sample rate. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 1 + sound/core/pcm_misc.c | 18 ++++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d58d79..18ad1f1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -892,6 +892,7 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9c9eff9..d4fc1bf 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -488,3 +488,21 @@ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) return SNDRV_PCM_RATE_KNOT; } EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit); + +/** + * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate + * @rate_bit: the rate bit to convert + * + * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag + * or 0 for an unknown rate bit + */ +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit) +{ + unsigned int i; + + for (i = 0; i < snd_pcm_known_rates.count; i++) + if ((1u << i) == rate_bit) + return snd_pcm_known_rates.list[i]; + return 0; +} +EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate); -- 1.7.10.4

2 2

[alsa-devel] [PATCH] snd-opti9xx: fixes for MED3931 card (opti931)
by Ondrej Zary 18 Jun '12

18 Jun '12

MED3931 card did not work (failed with "OPTI chip not found") because snd-opti9xx gets mc_indir_index from pnp by adding 2 to the pnp-reported port. It probably works for some cards but not for this one. Datasheet says that the port is always at 0xe?e so just force the lowest nibble to be 0xe. Also this card powers up with (ugly) 3D sound enabled. As there's no mixer control for this, just disable it. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,7 +135,6 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; - unsigned long mc_indir_size; struct resource *res_mc_indir; struct snd_wss *codec; #endif /* OPTi93X */ @@ -245,10 +244,8 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - if (!chip->mc_indir_index) { + if (!chip->mc_indir_index) chip->mc_indir_index = 0xe0e; - chip->mc_indir_size = 2; - } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -403,7 +400,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, #else /* OPTi93X */ case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: + /* disable 3D sound (set GPIO1 as output, low) */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); + case OPTi9XX_HW_82C933: /* FALL THROUGH */ /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. @@ -696,8 +695,7 @@ static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) return 0; #else /* OPTi93X */ - chip->res_mc_indir = request_region(chip->mc_indir_index, - chip->mc_indir_size, + chip->res_mc_indir = request_region(chip->mc_indir_index, 2, "OPTi93x MC"); if (chip->res_mc_indir == NULL) return -EBUSY; @@ -770,8 +768,9 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; - chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; - chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; + /* adjust mc_indir_index - some cards report it at 0xe?d, + other at 0xe?c but it really is always at 0xe?e */ + chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe; #else devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); if (devmc == NULL) -- Ondrej Zary

2 1

[alsa-devel] [PATCH] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 18 Jun '12

18 Jun '12

This addresses an issue encountered when a pcm is opened while transitioning to low power state (codec->power_on == 1 && codec->power_transition == -1). Add snd_pcm_power_up_d3wait to hda_codec. This function is used to power up from azx_open as opposed to snd_hda_power_up used from codec_exec_verb. When powering up from azx_open, wait for pending power downs to complete, avoiding the power up continuing in parallel with the power down on the work queue. The specific issue seen was with the CS4210 codec, it powers off the ADC and DAC nid in its suspend handler. If it is re-opened before the ~100ms power down process completes, the ADC and DAC nid are initialized while powered down and audio is lost until another suspend/resume cycle. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 46 ++++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_intel.c | 2 +- 3 files changed, 40 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 851e6ec..604699c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4423,20 +4423,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) +/* Transition to powered up, if wait_power_down then wait for a pending + * transition to D3 to complete. A pending D3 transition is indicated + * with power_transition == -1. */ +static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition > 0) { + /* Return if power_on or transitioning to power_on, unless currently + * powering down. */ + if ((codec->power_on || codec->power_transition > 0) && + !(wait_power_down && codec->power_transition < 0)) { spin_unlock(&codec->power_lock); return; } @@ -4460,8 +4459,37 @@ void snd_hda_power_up(struct hda_codec *codec) codec->power_transition = 0; spin_unlock(&codec->power_lock); } + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +void snd_hda_power_up(struct hda_codec *codec) +{ + __snd_hda_power_up(codec, false); +} EXPORT_SYMBOL_HDA(snd_hda_power_up); +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + /* This will cancel and wait for pending power_work to complete. */ + __snd_hda_power_up(codec, true); +} +EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 71864cdd..a4ac1de 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1059,10 +1059,12 @@ const char *snd_hda_get_jack_location(u32 cfg); */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8f85d30..86758dd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1767,7 +1767,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up(apcm->codec); + snd_hda_power_up_d3wait(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); -- 1.7.9.rc0

2 1

[alsa-devel] [PATCH] ALSA: snd-usb: fix sync pipe check
by Daniel Mack 18 Jun '12

18 Jun '12

Fix a bogus sanity check for sync pipe in pcm.c. This flaw was introduced during the streaming logic refactorization. While at it, improve the error messages that are generated in such cases. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: <ben(a)b1c1l1.com> --- sound/usb/pcm.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b76..fc7ce7c 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0 && !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || - ( is_playback && !implicit_fb))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); return -EINVAL; } -- 1.7.10.2

2 1

[alsa-devel] [PATCH] ALSA: 6fire: use NULL instead of 0 for pointer assignment
by Daniel Mack 18 Jun '12

18 Jun '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Torsten Schenk <torsten.schenk(a)zoho.com> --- sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 6f9715a..56ad923 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload( int ret; u8 data; struct usb_device *device = interface_to_usbdev(intf); - const struct firmware *fw = 0; + const struct firmware *fw = NULL; struct ihex_record *rec = kmalloc(sizeof(struct ihex_record), GFP_KERNEL); -- 1.7.10.2

2 1

[alsa-devel] [PATCH] ALSA: snd-usb: make snd_usb_substream_capture_trigger static
by Daniel Mack 18 Jun '12

18 Jun '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b76..ce22f5f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1147,7 +1147,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea return -EINVAL; } -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, + int cmd) { int err; struct snd_usb_substream *subs = substream->runtime->private_data; -- 1.7.10.2

2 1

[alsa-devel] ASoC updates for 3.5
by Mark Brown 18 Jun '12

18 Jun '12

The following changes since commit 69c5b7530651c8e375d2fbecd157a9a9a20c4cc9: ASoC: tegra: add MODULE_DEVICE_TABLE to tegra30_ahub (2012-06-08 08:49:30 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to 3419ae781f1592b3d367107db6500090495490cd: ASoC: tegra+wm8903: turn of mic detect when card is removed (2012-06-12 11:38:48 +0800) ---------------------------------------------------------------- ASoC: Updates for 3.5 All driver specific and fairly small. The pxa-ssp changes are larger than I'd like but they're build failures and are pretty clear to inspection. ---------------------------------------------------------------- Mark Brown (5): ASoC: wm8904: Fix GPIO and MICBIAS initialisation for regmap conversion ASoC: wm8904: Fix cache only management ASoC: wm8996: Remove spurious regulator_bulk_free() ASoC: wm8996: Move reset before the initial regulator disable ASoC: wm8996: Mark the CODEC as cache only when powering off on boot Qiao Zhou (2): ARM: MMP: add pxa910-ssp into ssp_id_table ASoC: fix pxa-ssp compiling issue under mach-mmp Stephen Warren (1): ASoC: tegra+wm8903: turn of mic detect when card is removed arch/arm/plat-pxa/ssp.c | 1 + include/linux/pxa2xx_ssp.h | 2 ++ include/linux/spi/pxa2xx_spi.h | 2 +- sound/soc/codecs/wm8904.c | 26 ++++++++++++-------------- sound/soc/codecs/wm8996.c | 8 +++----- sound/soc/pxa/pxa-ssp.c | 38 +++++++++++--------------------------- sound/soc/tegra/tegra_wm8903.c | 13 +++++++++++++ 7 files changed, 43 insertions(+), 47 deletions(-)

2 1

[alsa-devel] [patch] ASoC: Ux500: unlock on an error path
by Dan Carpenter 17 Jun '12

17 Jun '12

There is a missing mutex_unlock() here. The cleanup path also has more debug output. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 389dd66..3c79592 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1235,7 +1235,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", __func__, enum_anc_state[req]); - return -EINVAL; + status = -EINVAL; + goto cleanup; } apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;

2 1

[alsa-devel] [PATCH] ASoC: Ux500: Add machine-driver
by Ola Lilja 17 Jun '12

17 Jun '12

Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 11 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/mop500.c | 113 ++++++++++ sound/soc/ux500/mop500_ab8500.c | 431 +++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.h | 22 ++ 5 files changed, 580 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 1d38515..069330d 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -19,3 +19,14 @@ config SND_SOC_UX500_PLAT_DMA select SND_SOC_DMAENGINE_PCM help Say Y if you want to enable the Ux500 platform-driver. + ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 4634bf0..cce0c11 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -5,3 +5,6 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 0000000..31c4d26 --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,113 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + mop500_ab8500_remove(mop500_card); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 0000000..78cce23 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,431 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <mach/hardware.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Clocks */ +static const char * const enum_mclk[] = { + "SYSCLK", + "ULPCLK" +}; +enum mclk { + MCLK_SYSCLK, + MCLK_ULPCLK, +}; + +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk); + +/* Private data for machine-part MOP500<->AB8500 */ +struct mop500_ab8500_drvdata { + /* Clocks */ + enum mclk mclk_sel; + struct clk *clk_ptr_intclk; + struct clk *clk_ptr_sysclk; + struct clk *clk_ptr_ulpclk; +}; + +static inline const char *get_mclk_str(enum mclk mclk_sel) +{ + switch (mclk_sel) { + case MCLK_SYSCLK: + return "SYSCLK"; + case MCLK_ULPCLK: + return "ULPCLK"; + default: + return "Unknown"; + } +} + +static int mop500_ab8500_set_mclk(struct device *dev, + struct mop500_ab8500_drvdata *drvdata) +{ + int status; + struct clk *clk_ptr; + + if (IS_ERR(drvdata->clk_ptr_intclk)) { + dev_err(dev, + "%s: ERROR: intclk not initialized!\n", __func__); + return -EIO; + } + + switch (drvdata->mclk_sel) { + case MCLK_SYSCLK: + clk_ptr = drvdata->clk_ptr_sysclk; + break; + case MCLK_ULPCLK: + clk_ptr = drvdata->clk_ptr_ulpclk; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clk_ptr)) { + dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__, + get_mclk_str(drvdata->mclk_sel)); + return -EIO; + } + + status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr); + if (status) + dev_err(dev, + "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!", + __func__, get_mclk_str(drvdata->mclk_sel), status); + else + dev_dbg(dev, + "%s: intclk parent changed to %s.\n", + __func__, get_mclk_str(drvdata->mclk_sel)); + + return status; +} + +/* + * Control-events + */ + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + + ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; + + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + unsigned int val = ucontrol->value.enumerated.item[0]; + + if (val > (unsigned int)MCLK_ULPCLK) + return -EINVAL; + if (drvdata->mclk_sel == val) + return 0; + + drvdata->mclk_sel = val; + + return 1; +} + +/* + * Controls + */ + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_ENUM_EXT("Master Clock Select", + soc_enum_mclk, + mclk_input_control_get, mclk_input_control_put), + /* Digital interface - Clocks */ + SOC_SINGLE("Digital Interface Master Generator Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, + 1, 0), + SOC_SINGLE("Digital Interface 0 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, + 1, 0), + SOC_SINGLE("Digital Interface 1 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, + 1, 0), + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +int mop500_ab8500_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set audio-clock source */ + return mop500_ab8500_set_mclk(rtd->card->dev, + snd_soc_card_get_drvdata(rtd->card)); +} + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct device *dev = rtd->card->dev; + unsigned int fmt; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + if (channels == 8) + driver_mode = DRIVERMODE_CODEC_ONLY; + else + driver_mode = DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + if (driver_mode == DRIVERMODE_NORMAL) + sw_codec = sw_cpu; + else + sw_codec = 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .startup = mop500_ab8500_startup, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_ab8500_drvdata *drvdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), + GFP_KERNEL); + snd_soc_card_set_drvdata(rtd->card, drvdata); + + /* Setup clocks */ + + drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk"); + if (IS_ERR(drvdata->clk_ptr_sysclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n", + __func__); + drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk"); + if (IS_ERR(drvdata->clk_ptr_ulpclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n", + __func__); + drvdata->clk_ptr_intclk = clk_get(dev, "intclk"); + if (IS_ERR(drvdata->clk_ptr_intclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n", + __func__); + + /* Set intclk default parent to ulpclk */ + drvdata->mclk_sel = MCLK_ULPCLK; + ret = mop500_ab8500_set_mclk(dev, drvdata); + if (ret < 0) + dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n", + __func__); + + drvdata->mclk_sel = MCLK_ULPCLK; + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + +void mop500_ab8500_remove(struct snd_soc_card *card) +{ + struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (drvdata->clk_ptr_sysclk != NULL) + clk_put(drvdata->clk_ptr_sysclk); + if (drvdata->clk_ptr_ulpclk != NULL) + clk_put(drvdata->clk_ptr_ulpclk); + if (drvdata->clk_ptr_intclk != NULL) + clk_put(drvdata->clk_ptr_intclk); + + snd_soc_card_set_drvdata(card, drvdata); +} diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 0000000..cca5b33 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,22 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); +void mop500_ab8500_remove(struct snd_soc_card *card); + +#endif -- 1.7.8.3

3 2

[alsa-devel] [PATCH] USB: fix gathering of interface associations
by Daniel Mack 17 Jun '12

17 Jun '12

TEAC's UD-H01 (and probably other devices) have a gap in the interface number allocation of their descriptors: Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 220 bNumInterfaces 3 [...] Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 [...] Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 [...] Once a configuration is selected, usb_set_configuration() walks the known interfaces of a given configuration and calls find_iad() on each of them to set the interface association pointer the interface is included in. The problem here is that the loop variable is taken for the interface number in the comparison logic that gathers the association. Which is fine as long as the descriptors are sane. In the case above, however, the logic gets out of sync and the interface association fields of all interfaces beyond the interface number gap are wrong. Fix this by passing the interface's bInterfaceNumber to find_iad() instead. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: bEN <ml_all(a)circa.be> Reported-by: Ivan Perrone <ivanperrone(a)hotmail.com> --- This patch has been tested on a virtual USB device emulated in QEMU. I'm still waiting for test results based on real hardware, but the bug is rather obvious is and the fix really simple. Daniel drivers/usb/core/message.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/usb/core/message.c b/drivers/usb/core/message.c index b548cf1..bdd1c67 100644 --- a/drivers/usb/core/message.c +++ b/drivers/usb/core/message.c @@ -1838,7 +1838,6 @@ free_interfaces: intfc = cp->intf_cache[i]; intf->altsetting = intfc->altsetting; intf->num_altsetting = intfc->num_altsetting; - intf->intf_assoc = find_iad(dev, cp, i); kref_get(&intfc->ref); alt = usb_altnum_to_altsetting(intf, 0); @@ -1851,6 +1850,8 @@ free_interfaces: if (!alt) alt = &intf->altsetting[0]; + intf->intf_assoc = + find_iad(dev, cp, alt->desc.bInterfaceNumber); intf->cur_altsetting = alt; usb_enable_interface(dev, intf, true); intf->dev.parent = &dev->dev; -- 1.7.10.2

2 3

[alsa-devel] [PATCH 0/2] RFC: support for audio wall clock
by Pierre-Louis Bossart 15 Jun '12

15 Jun '12

Attached a set of patches to show how the audio wall clock (eg. the HDAudio one) can be used to provide precise audio timestamps and track the drift between audio time and system time. On my laptop I am able to track a 7ppm delta. I will explain in more details how this can be used during LPC in San Diego, but with the summer coming I'd like to get feedback as there are changes to the core/alsa-lib that haven't been modified in years. Pierre-Louis Bossart (2): ALSA: core: add hooks for audio timestamps read from WALLCLOCK ALSA: hda: support for wallclock timestamps include/sound/asound.h | 7 +++- include/sound/pcm.h | 2 + sound/core/pcm_lib.c | 15 ++++++- sound/core/pcm_native.c | 2 + sound/pci/hda/hda_intel.c | 106 ++++++++++++++++++++++++++++++++++++++++++++- 5 files changed, 128 insertions(+), 4 deletions(-) -- 1.7.6.5

4 11

[alsa-devel] [PATCH] pcm: support for audio wallclock
by Pierre-Louis Bossart 15 Jun '12

15 Jun '12

add new snd_pcm_status_get_htstamp_wallclk() routine to query the audio timestamps read from the wallclock. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 1 + include/sound/asound.h | 7 ++++++- src/pcm/pcm.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+), 1 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..325c0ea 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -858,6 +858,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..c186fd6 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,10 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + union { + unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ + } ext; }; struct sndrv_pcm_mmap_status { @@ -434,6 +438,7 @@ struct sndrv_pcm_mmap_status { int pad1; /* Needed for 64 bit alignment */ sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ int suspended_state; /* RO: suspended stream state */ }; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..2221865 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -6213,6 +6213,22 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio wallclock timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +#ifndef DOXYGEN +void INTERNAL(snd_pcm_status_get_htstamp_wallclk)(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +#else +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +#endif +{ + assert(obj && ptr); + *ptr = obj->ext.audio_tstamp; +} +use_default_symbol_version(__snd_pcm_status_get_htstamp_wallclk, snd_pcm_status_get_htstamp_wallclk, ALSA_0.9.0rc8); /* FIXME, no idea why this is needed */ + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * -- 1.7.6.5

2 2

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Convert to use devm_*
by Peter Ujfalusi 15 Jun '12

15 Jun '12

Switch to use devm_* te make the probe/remove code more cleaner. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 52 ++++++++++-------------------------------- 1 files changed, 13 insertions(+), 39 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 59d47ab..b5e23af 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -445,9 +445,8 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { struct omap_mcpdm *mcpdm; struct resource *res; - int ret = 0; - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL); if (!mcpdm) return -ENOMEM; @@ -456,55 +455,30 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) mutex_init(&mcpdm->mutex); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_res; - } + if (res == NULL) + return -ENOMEM; - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_res; - } + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), "McPDM")) + return -EBUSY; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_iomap; - } + mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!mcpdm->io_base) + return -ENOMEM; mcpdm->irq = platform_get_irq(pdev, 0); - if (mcpdm->irq < 0) { - ret = mcpdm->irq; - goto err_irq; - } + if (mcpdm->irq < 0) + return mcpdm->irq; mcpdm->dev = &pdev->dev; - ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (!ret) - return 0; - -err_irq: - iounmap(mcpdm->io_base); -err_iomap: - release_mem_region(res->start, resource_size(res)); -err_res: - kfree(mcpdm); - return ret; + return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); - struct resource *res; - snd_soc_unregister_dai(&pdev->dev); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - iounmap(mcpdm->io_base); - release_mem_region(res->start, resource_size(res)); - - kfree(mcpdm); return 0; } -- 1.7.8.6

2 1

Re: [alsa-devel] [PATCH 1/1] hda_intel: activate COMBO mode for Intel client chipsets
by Takashi Iwai 15 Jun '12

15 Jun '12

At Thu, 14 Jun 2012 14:23:53 -0700, Seth Heasley wrote: > > This patch activates the COMBO position_fix for recent Intel client chipsets. COMBO mode is the recommended setting for Intel chipsets and eliminates HD audio warnings in dmesg. This patch has been tested on Lynx Point, Panther Point, and Cougar Pont. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > sound/pci/hda/hda_intel.c | 11 ++++++++--- > 1 files changed, 8 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 2b6392b..466a88c 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -535,6 +535,7 @@ enum { > #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ > #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ > #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ > +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ > > /* quirks for ATI SB / AMD Hudson */ > #define AZX_DCAPS_PRESET_ATI_SB \ > @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) > snd_printd(SFX "Using LPIB position fix\n"); > return POS_FIX_LPIB; > } > + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > + snd_printd(SFX "Using COMBO position fix\n"); > + return POS_FIX_COMBO; > + } > return POS_FIX_AUTO; > } > > @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE }, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > @@ -3248,11 +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Panther Point */ > { PCI_DEVICE(0x8086, 0x1e20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), > .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > -- > 1.7.4.4 >

1 0

[alsa-devel] McBSP in TDM for audio codec
by learnom learnom 15 Jun '12

15 Jun '12

Hi All, I have to support 4 channels in I2S format in TDM mode. Processor: OMAP 3703 If any one have any idea on how to go about this. Please let me know. Thanks, learnom

2 3

[alsa-devel] Anyone using alsactl preinit / postinit scripts?
by Takashi Iwai 14 Jun '12

14 Jun '12

Hi, is anyone already using preinit / postinit scripts of alsactl init? I'm thinking of appliny the patch below, but before that, I'd like to make sure that *.conf is the appropriate extension. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] alsactl: Read only *.conf files when a directory is passed via INCLUDE When alsactl init is invoked and a directory path is passed to INCLUDE command in the config file, read only *.conf files in that directory. This will avoid reading backup files or invalid files that have been created accidentally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/init_parse.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/alsactl/init_parse.c b/alsactl/init_parse.c index 51b515c..83d12ba 100644 --- a/alsactl/init_parse.c +++ b/alsactl/init_parse.c @@ -1502,12 +1502,16 @@ static int parse_line(struct space *space, char *line, size_t linesize) if (dir) { count = strlen(string); while ((dirent = readdir(dir)) != NULL) { - if (strcmp(dirent->d_name, ".") == 0 || - strcmp(dirent->d_name, "..") == 0) + const char *name = dirent->d_name; + const char *ext; + if (*name == '.') + continue; + ext = strrchr(name, ','); + if (!ext || strcmp(ext, ".conf")) continue; string[count] = '\0'; strlcat(string, "/", sizeof(string)); - strlcat(string, dirent->d_name, sizeof(string)); + strlcat(string, name, sizeof(string)); space->go_to = NULL; space->rootdir = new_root_dir(string); if (space->rootdir) { -- 1.7.10.3

2 2

[alsa-devel] [PATCH] ALSA: HDA: Create phantom jacks for fixed inputs and outputs
by David Henningsson 14 Jun '12

14 Jun '12

PulseAudio sometimes have difficulties knowing that there is a "Speaker" or "Internal Mic", if they have no individual volume controls or selectors. As a result, only e g "Headphone" might be created for a laptop, but no "Speaker". To help out, create phantom jacks (that are always present, at least for now) for "Speaker", "Internal Mic" etc, in case we detect them. The naming convention is e g "Speaker Phantom Jack". In order not to pollute the /dev/input namespace with even more devices, these are added to the kcontrols only, not the input devices. An implementation note: The possibility to override how to actually do jack sensing opens up for more advanced usage of the jack infrastructure in the future (e g letting the headphone jack detection set presence status for both headphone and mic in case of a headset jack, among other things). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 64 ++++++++++++++++++++++++++++++++++------------ sound/pci/hda/hda_jack.h | 5 ++++ 2 files changed, 52 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 2dd1c11..c5569f1 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -133,6 +133,13 @@ static void jack_detect_update(struct hda_codec *codec, } } +static void jack_detect_always_present(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + jack->pin_sense = AC_PINSENSE_PRESENCE; + jack->jack_dirty = 0; +} + /** * snd_hda_set_dirty_all - Mark all the cached as dirty * @@ -162,7 +169,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); if (jack) { - jack_detect_update(codec, jack); + if (jack->update_func) + jack->update_func(codec, jack); return jack->pin_sense; } return read_pin_sense(codec, nid); @@ -215,7 +223,8 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) for (i = 0; i < codec->jacktbl.used; i++, jack++) if (jack->nid) { - jack_detect_update(codec, jack); + if (jack->update_func) + jack->update_func(codec, jack); if (!jack->kctl) continue; state = get_jack_plug_state(jack->pin_sense); @@ -264,8 +273,8 @@ static void hda_free_jack_priv(struct snd_jack *jack) * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) +static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx, bool phantom_jack) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -283,19 +292,33 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (err < 0) return err; jack->kctl = kctl; + if (phantom_jack) + jack->update_func = jack_detect_always_present; + else + jack->update_func = jack_detect_update; + state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK - jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - snd_jack_report(jack->jack, state ? jack->type : 0); + if (!phantom_jack) { + jack->type = get_input_jack_type(codec, nid); + err = snd_jack_new(codec->bus->card, name, jack->type, + &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + snd_jack_report(jack->jack, state ? jack->type : 0); + } #endif return 0; } + +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx) +{ + return __snd_hda_jack_add_kctl(codec, nid, name, idx, false); +} EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, @@ -305,25 +328,32 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, unsigned int def_conf, conn; char name[44]; int idx, err; + bool phantom_jack; if (!nid) return 0; - if (!is_jack_detectable(codec, nid)) - return 0; def_conf = snd_hda_codec_get_pincfg(codec, nid); conn = get_defcfg_connect(def_conf); - if (conn != AC_JACK_PORT_COMPLEX) + if (conn == AC_JACK_PORT_NONE) return 0; + phantom_jack = (conn != AC_JACK_PORT_COMPLEX) || + !is_jack_detectable(codec, nid); snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (phantom_jack) + /* Example final name: "Internal Mic Phantom Jack" */ + strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); if (!strcmp(name, lastname) && idx == *lastidx) idx++; - strncpy(lastname, name, 44); + strncpy(lastname, name, sizeof(name)); *lastidx = idx; - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack); if (err < 0) return err; - return snd_hda_jack_detect_enable(codec, nid, 0); + + if (!phantom_jack) + return snd_hda_jack_detect_enable(codec, nid, 0); + return 0; } /** diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ae5246..4055236 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,6 +13,10 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*snd_hda_jack_detect_update)(struct hda_codec *codec, + struct hda_jack_tbl *jack); struct hda_jack_tbl { hda_nid_t nid; @@ -24,6 +28,7 @@ struct hda_jack_tbl { unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ + snd_hda_jack_detect_update update_func; /* function to call to perform jack detection */ #ifdef CONFIG_SND_HDA_INPUT_JACK int type; struct snd_jack *jack; -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/1] hda_intel: switch Intel chipsets to COMBO position_fix
by Takashi Iwai 14 Jun '12

14 Jun '12

At Wed, 13 Jun 2012 17:58:57 +0000, Heasley, Seth wrote: > > >> This patch implements the COMBO position_fix for recent Intel client > >chipsets. > >> > >> Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> > > > >Thanks for the patch. > >But could you give a bit more context why this change is needed? > > The COMBO method was recommended for Intel chipsets by a certain audio maintainer. ;) It was also tested on the most recent Intel chips because we were seeing warning messages when using DMA positioning ("Invalid position buffer, using LPIB read method instead."). Could you resend the patch with such some reasonable comments in the changelog? thanks, Takashi > >> --- > >> sound/pci/hda/hda_intel.c | 11 ++++++++--- > >> 1 files changed, 8 insertions(+), 3 deletions(-) > >> > >> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > >> index 2b6392b..466a88c 100644 > >> --- a/sound/pci/hda/hda_intel.c > >> +++ b/sound/pci/hda/hda_intel.c > >> @@ -535,6 +535,7 @@ enum { > >> #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size > >alignment */ > >> #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size > >alignment */ > >> #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary > >*/ > >> +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as > >default */ > >> > >> /* quirks for ATI SB / AMD Hudson */ > >> #define AZX_DCAPS_PRESET_ATI_SB \ > >> @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx > >*chip, int fix) > >> snd_printd(SFX "Using LPIB position fix\n"); > >> return POS_FIX_LPIB; > >> } > >> + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > >> + snd_printd(SFX "Using COMBO position fix\n"); > >> + return POS_FIX_COMBO; > >> + } > >> return POS_FIX_AUTO; > >> } > >> > >> @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > >> /* CPT */ > >> { PCI_DEVICE(0x8086, 0x1c20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > >> - AZX_DCAPS_BUFSIZE }, > >> + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > >> /* PBG */ > >> { PCI_DEVICE(0x8086, 0x1d20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | @@ - > >3248,11 > >> +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > >> /* Panther Point */ > >> { PCI_DEVICE(0x8086, 0x1e20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > >> - AZX_DCAPS_BUFSIZE}, > >> + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > >> /* Lynx Point */ > >> { PCI_DEVICE(0x8086, 0x8c20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > >> - AZX_DCAPS_BUFSIZE}, > >> + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > >> /* SCH */ > >> { PCI_DEVICE(0x8086, 0x811b), > >> .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > >> -- > >> 1.7.4.4 > >> >

1 0

[alsa-devel] Does amixer have commands to turn on / off Codec TWL4030 analog microphone bias voltage output?
by Philip Chu 14 Jun '12

14 Jun '12

I am using Linux kernel 3.0 and a Logic PD DM3530 SOM LV, looking for the amixer commands to turn on / off analog microphone bias supply. Following lines do not have the bias needed for Headset MIC input: amixer set 'Analog Left AUXL' nocap amixer set 'Analog Right AUXR' nocap amixer set 'Analog Left Main Mic' nocap amixer set 'Analog Left Headset Mic' cap Anybody can help? Thanks, Philip Chu principal software engineer LogicPD www.logicpd.com 5 Clock Tower Place, Suite 215 Maynard, MA 01754 Phone: 978-243-2450 Fax: 978-461-1598

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Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by Arnd Bergmann 14 Jun '12

14 Jun '12

On Monday 11 June 2012, Zhangfei Gao wrote: > > v1->v2: > modify accroingly with suggestion from Vinod and Mark > remove deprecated chan->private and dma.h > remove prep_slave_sg > remove module_init and move operation to probe > misc change It seems that this driver is missing DT bindings, which really need to get added. The brownstone platform is currently supported using both DT and ATAG probing, but we want to remove the ATAG variant eventually, so adding a new driver for that platform without support for DT booting is a step in the wrong direction. Arnd -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

Re: [alsa-devel] Questions regarding adding regmap support for TLV320AIC3262
by Mark Brown 13 Jun '12

13 Jun '12

On Tue, Jun 12, 2012 at 08:45:51AM +0000, Navada Kanyana, Mukund wrote: Fix your mailer to word wrap within paragraphs. See email-clients.txt in the kernel. > In order to support such a regmap, I would like to push the > following patch for regmap.c to add support for 24-bit register > addressing. This looks fine, please submit as covered in SubmittingPatches in the kernel source. You might also want to have a look in the list archives for the patches that some of the guys at Dialog have been submitting for devices with paged register maps, it sounds like these might be useful for this device. Please always CC the appropriate mailing list for the subsytem, in this case linux-kernel. There's nothing ASoC specific about this.

1 0

[alsa-devel] [PATCH 1/5] ASoC: wm8903: Move pin configuration into I2C probe() function
by Mark Brown 13 Jun '12

13 Jun '12

Ensure that the device pins are configured as soon as possible by moving the pin configration (including MICBIAS) into the I2C probe() function. This had been done in the CODEC probe() function when we were relying on the ASoC register I/O code. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This series is untested but should deal with the WM8903 side of what I was saying earlier. sound/soc/codecs/wm8903.c | 92 +++++++++++++++++++++++---------------------- 1 file changed, 47 insertions(+), 45 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e76a66d..7200088 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1,7 +1,7 @@ /* * wm8903.c -- WM8903 ALSA SoC Audio driver * - * Copyright 2008-11 Wolfson Microelectronics + * Copyright 2008-12 Wolfson Microelectronics * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> @@ -2081,10 +2081,9 @@ static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); struct wm8903_platform_data *pdata = wm8903->pdata; - int ret, i; + int ret; int trigger, irq_pol; u16 val; - bool mic_gpio = false; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -2095,47 +2094,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - /* Set up GPIOs, detect if any are MIC detect outputs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if ((!pdata->gpio_cfg[i]) || - (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) - continue; - - snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0x7fff); - - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) - >> WM8903_GP1_FN_SHIFT; - - switch (val) { - case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: - case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: - mic_gpio = true; - break; - default: - break; - } - } - - /* Set up microphone detection */ - snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); - - /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) - snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, - WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); - - /* If microphone detection is enabled by pdata but - * detected via IRQ then interrupts can be lost before - * the machine driver has set up microphone detection - * IRQs as the IRQs are clear on read. The detection - * will be enabled when the machine driver configures. - */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); - - wm8903->mic_delay = pdata->micdet_delay; - if (wm8903->irq) { if (pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; @@ -2316,8 +2274,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, { struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; + bool mic_gpio = false; unsigned int val; - int ret; + int ret, i; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), GFP_KERNEL); @@ -2361,6 +2320,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } } + pdata = wm8903->pdata; + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); @@ -2385,6 +2346,47 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, wm8903_init_gpio(wm8903); + /* Set up GPIO pin state, detect if any are MIC detect outputs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) + continue; + + regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0x7fff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } + } + + /* Set up microphone detection */ + regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); + + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + + wm8903->mic_delay = pdata->micdet_delay; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) -- 1.7.10

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Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by Mark Brown 13 Jun '12

13 Jun '12

On Mon, Jun 11, 2012 at 06:04:36PM +0800, Zhangfei Gao wrote: > These patches provide mmp audio support under alsa via dmaengine > Support platfrom pxa688 and pxa910 > mmp_tdma.c and mmp-pcm.c is shared for pxa688 and pxa910 > pxa688 (mmp2) use mmp2-sspa.c > pxa910 directly use pxa-ssp.c > mmp_tdma.c is under dmaengine framework Applied 2-4. I'm not seeing a direct code dependency between this and the dmaengine driver in patch 1 (or review of it from Vinod yet) so I'm expecting that to go via the dmaengine tree.

2 2

[alsa-devel] Possible message repeat
by Brent Weatherall 13 Jun '12

13 Jun '12

Hello. I sent this message previously before I solidified my membership to the list and it seems my message never posted. Here it is again. My apologies if did go through and I missed it. I am new to audio development in Linux. I have worked with user space development with respect to UVC code and I have worked with specifying an XU in UVC. I wrote a user space driver/application to talk to a device supporting the UVC XU in Linux. I am trying to specify an extension to the UAC specification and then write a user space driver/application to use the extension when a device that supports it is available in Linux. Pretty much the same situation as I have already done with UVC XU. Windows and Mac code will follow, but right now I am only concerned with Linux. Can anyone assist me with the best path forward for writing user space code to use the XU I am going to define? Meaning generically speaking how to use UAC XU controls in ALSA. I was looking for code samples that use the audio.h header file interface, but after a day and a half I am coming to conclusion that only kernel space uses audio.h and that I will need to go through ALSA to accomplish my task. Again, this will be user space code I want distribute easily, with minimal dependencies. Our code will be apache licensed afterwards if possible, just like our UVC code was. So any work I do should be freely available to the community afterwards. **EDIT FROM PREVIOUS MESSAGE** After more searching, it appears that any XU controls would be processed through usbmixer.c, but I don't see how to use the code in usbmixer from user space. And so far I can't tell what source usbmixer resides in. I downloaded the alsa-lib source in Fedora, but I don't see that file. Is this a Linux kernel file? So far I am having lots of trouble finding anything useful regarding audio XUs in Linux. My apologies if this should be simple. Thank you in advance for any assistance. Best Regards, Brent

2 1

[alsa-devel] [PATCH 1/4] ASoC: when initializing CPU DAI, don't duplicate any CODEC init
by Stephen Warren 13 Jun '12

13 Jun '12

From: Stephen Warren <swarren(a)nvidia.com> If the CPU-side of a DAI link is a CODEC rather than a standalone DAI, the codec initialization will call try_module_get() and create the DAI widgets. Ensure that this isn't duplicated when the CPU DAI itself is probed, if the CPU DAI is part of a CODEC. Note that this is not a complete fix on its own, since there's no guarantee that the CODEC itself will be initialized - currently that only happens if the CODEC is also used as the CODEC-side of a DAI link, and that initialization may happen before or after the DAIs within the CODEC are initialized. However, such a scenario doesn't necessarily currently work, and I don't think this change alone makes it any worse. This is fixed in a couple patches time. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 14 +++++++++----- 1 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3d803f3..448d4a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -983,7 +983,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); - module_put(cpu_dai->dev->driver->owner); + + if (!cpu_dai->codec) + module_put(cpu_dai->dev->driver->owner); } } @@ -1257,11 +1259,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - cpu_dai->dapm.card = card; - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; + if (!cpu_dai->codec) { + cpu_dai->dapm.card = card; + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + } if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); -- 1.7.0.4

2 4

Re: [alsa-devel] [PATCH v3] sound/soc/lapis: add platform driver for ML7213
by Mark Brown 13 Jun '12

13 Jun '12

On Tue, Mar 06, 2012 at 02:48:05PM +0900, Tomoya MORINAGA wrote: > 2012年3月3日1:39 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > I just merged Lars-Peter's dmaengine library code which has been on the > > list for a week or so - this should be updated to use that next time > > it's posted. That should save a lot of code from the driver and make > > sure it's following best practices for dmaengine use. > Sorry, we can't use this library. This sort of response really isn't on. Do you have some reason for saying this? Flat out refusing to do things with no reason is not useful. > >> +static struct ioh_i2s_data *i2s_data; > >> +static struct ioh_i2s_dma dmadata[MAX_I2S_CH]; > > Why are these needed, aren't they dynamically allocated by the driver? > You mean ASoC driver can't use global variable ? In general no Linux driver should be using a global variable except for things like this. > >> + case SNDRV_PCM_FORMAT_S32_LE: > >> + byte = 24; > >> + break; > > That looks wrong... are you sure you don't support S24_LE or something? > This is correct. > Because maximum transmit size of ML7213's I2S hw is 24bit. Note we often lay out 24 bit audio in 32 bit blocks. > >> +static struct platform_driver ioh_i2s_driver_plat = { > > > >> +static struct platform_driver ioh_dai_driver_plat = { > > > >> +static int ioh_i2s_pci_probe(struct pci_dev *pdev, > >> + const struct pci_device_id *id) > > Why are you creating these platform devices? I don't understand the > > function they serve. The code handling them looks to have quite a few > > problems but I'm not clear they should be there in the first place. > if these platform devices aren't used, device detection doesn't work correctly. > So, I added these. You've not actually mentioned the problem you were seeing... > >> + rv = request_irq(pdev->irq, ioh_i2s_irq, IRQF_SHARED, "ml7213_ioh", > >> + pdev); > >> + if (rv != 0) { > >> + printk(KERN_ERR "Failed to allocate irq\n"); > >> + goto out_irq; > >> + } > > Are you *sure* you're ready to handle interrupts at this point? > This is just registering interrupt handler. > As long as interrupt register is not enabled, the interrupt handler is > not called. > This is common request_irq description. > What's is your concern ? Apart from anything else you've got the interrupt requested as IRQF_SHARED so the interrupt could get called at any time. It's also not clear that you've got the hardware in a known good state.

3 28

[alsa-devel] [PATCH] ASoC: ml26124: Convert to devm_regmap_init_i2c
by Axel Lin 13 Jun '12

13 Jun '12

This fixes a leak if snd_soc_register_codec fails. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ml26124.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 22cb5bf..96aa5fa 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -638,7 +638,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, priv); - priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); @@ -651,10 +651,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, static __devexit int ml26124_i2c_remove(struct i2c_client *client) { - struct ml26124_priv *priv = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(priv->regmap); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Ux500: Correct license strings
by Ola Lilja 13 Jun '12

13 Jun '12

GPLv2 -> GPL v2 Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_pcm.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index dba246a..dd419c8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2510,4 +2510,4 @@ static struct platform_driver ab8500_codec_platform_driver = { }; module_platform_driver(ab8500_codec_platform_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 93c6c40..62ac028 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -840,4 +840,4 @@ static struct platform_driver msp_i2s_driver = { }; module_platform_driver(msp_i2s_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 496dec1..ee14d2d 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -739,4 +739,4 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, devm_kfree(&pdev->dev, msp); } -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 66b080e..97d8e4d 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -315,4 +315,4 @@ static struct platform_driver ux500_pcm_driver = { }; module_platform_driver(ux500_pcm_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 1.7.8.3

3 2

Re: [alsa-devel] [PATCH 1/1] hda_intel: switch Intel chipsets to COMBO position_fix
by Takashi Iwai 13 Jun '12

13 Jun '12

At Tue, 12 Jun 2012 16:30:00 -0700, Seth Heasley wrote: > > This patch implements the COMBO position_fix for recent Intel client chipsets. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Thanks for the patch. But could you give a bit more context why this change is needed? Takashi > --- > sound/pci/hda/hda_intel.c | 11 ++++++++--- > 1 files changed, 8 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 2b6392b..466a88c 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -535,6 +535,7 @@ enum { > #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ > #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ > #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ > +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ > > /* quirks for ATI SB / AMD Hudson */ > #define AZX_DCAPS_PRESET_ATI_SB \ > @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) > snd_printd(SFX "Using LPIB position fix\n"); > return POS_FIX_LPIB; > } > + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > + snd_printd(SFX "Using COMBO position fix\n"); > + return POS_FIX_COMBO; > + } > return POS_FIX_AUTO; > } > > @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE }, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > @@ -3248,11 +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Panther Point */ > { PCI_DEVICE(0x8086, 0x1e20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), > .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > -- > 1.7.4.4 >

1 0

[alsa-devel] [PATCH] ALSA: HDA: Pin fixup for Zotac Z68 motherboard
by David Henningsson 12 Jun '12

12 Jun '12

Pin 0x1b was connected to the front panel connector, which according to the HDA standard should contain a mic and a headphone. In this case, the headphone was listed as "line out" by BIOS. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/993162 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 224410e..b896681 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6439,6 +6439,7 @@ enum { ALC662_FIXUP_ASUS_MODE7, ALC662_FIXUP_ASUS_MODE8, ALC662_FIXUP_NO_JACK_DETECT, + ALC662_FIXUP_ZOTAC_Z68, }; static const struct alc_fixup alc662_fixups[] = { @@ -6588,6 +6589,13 @@ static const struct alc_fixup alc662_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc_fixup_no_jack_detect, }, + [ALC662_FIXUP_ZOTAC_Z68] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1b, 0x02214020 }, /* Front HP */ + { } + } + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6601,6 +6609,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), 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(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), #if 0 -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] alsa: compress_core: don't wake up on pause
by Vinod Koul 12 Jun '12

12 Jun '12

during pause the core should maintain the status-quo on the device and pointers and not wake up. If app needs it should call DROP explcitly. Signed-off-by: Namarta Kohli <namartax.kohli(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/core/compress_offload.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a68aed7..375f7a0 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -502,10 +502,8 @@ static int snd_compr_pause(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) { + if (!retval) stream->runtime->state = SNDRV_PCM_STATE_PAUSED; - wake_up(&stream->runtime->sleep); - } return retval; } -- 1.7.0.4

2 2

[alsa-devel] [PATCH 3/3] mfd/ab8500: Move platform-data for ab8500-codec into mfd-driver
by Ola Lilja 12 Jun '12

12 Jun '12

The platform-data used by the Ux500 ASoC-driver is moved from the machine-driver context into the codec-driver context. This means adding the platform-data for 'ab8500-codec' into the main AB8500 platform-data. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/board-mop500.c | 14 ++++++++ include/linux/mfd/abx500/ab8500-codec.h | 52 +++++++++++++++++++++++++++++++ include/linux/mfd/abx500/ab8500.h | 2 + 3 files changed, 68 insertions(+), 0 deletions(-) create mode 100644 include/linux/mfd/abx500/ab8500-codec.h diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index c41912c..679a354 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -24,6 +24,7 @@ #include <linux/mfd/tc3589x.h> #include <linux/mfd/tps6105x.h> #include <linux/mfd/abx500/ab8500-gpio.h> +#include <linux/mfd/abx500/ab8500-codec.h> #include <linux/leds-lp5521.h> #include <linux/input.h> #include <linux/smsc911x.h> @@ -96,6 +97,18 @@ static struct ab8500_gpio_platform_data ab8500_gpio_pdata = { 0x7A, 0x00, 0x00}, }; +/* ab8500-codec */ +static struct ab8500_codec_platform_data ab8500_codec_pdata = { + .amics = { + .mic1_type = AMIC_TYPE_DIFFERENTIAL, + .mic2_type = AMIC_TYPE_DIFFERENTIAL, + .mic1a_micbias = AMIC_MICBIAS_VAMIC1, + .mic1b_micbias = AMIC_MICBIAS_VAMIC1, + .mic2_micbias = AMIC_MICBIAS_VAMIC2 + }, + .ear_cmv = EAR_CMV_0_95V +}; + static struct gpio_keys_button snowball_key_array[] = { { .gpio = 32, @@ -194,6 +207,7 @@ static struct ab8500_platform_data ab8500_platdata = { .regulator = ab8500_regulators, .num_regulator = ARRAY_SIZE(ab8500_regulators), .gpio = &ab8500_gpio_pdata, + .codec = &ab8500_codec_pdata, }; static struct resource ab8500_resources[] = { diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h new file mode 100644 index 0000000..dc65292 --- /dev/null +++ b/include/linux/mfd/abx500/ab8500-codec.h @@ -0,0 +1,52 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef AB8500_CORE_CODEC_H +#define AB8500_CORE_CODEC_H + +/* Mic-types */ +enum amic_type { + AMIC_TYPE_SINGLE_ENDED, + AMIC_TYPE_DIFFERENTIAL +}; + +/* Mic-biases */ +enum amic_micbias { + AMIC_MICBIAS_VAMIC1, + AMIC_MICBIAS_VAMIC2 +}; + +/* Bias-voltage */ +enum ear_cm_voltage { + EAR_CMV_0_95V, + EAR_CMV_1_10V, + EAR_CMV_1_27V, + EAR_CMV_1_58V +}; + +/* Analog microphone settings */ +struct amic_settings { + enum amic_type mic1_type; + enum amic_type mic2_type; + enum amic_micbias mic1a_micbias; + enum amic_micbias mic1b_micbias; + enum amic_micbias mic2_micbias; +}; + +/* Platform data structure for the audio-parts of the AB8500 */ +struct ab8500_codec_platform_data { + struct amic_settings amics; + enum ear_cm_voltage ear_cmv; +}; + +#endif diff --git a/include/linux/mfd/abx500/ab8500.h b/include/linux/mfd/abx500/ab8500.h index 91dd3ef..bc9b84b 100644 --- a/include/linux/mfd/abx500/ab8500.h +++ b/include/linux/mfd/abx500/ab8500.h @@ -266,6 +266,7 @@ struct ab8500 { struct regulator_reg_init; struct regulator_init_data; struct ab8500_gpio_platform_data; +struct ab8500_codec_platform_data; /** * struct ab8500_platform_data - AB8500 platform data @@ -284,6 +285,7 @@ struct ab8500_platform_data { int num_regulator; struct regulator_init_data *regulator; struct ab8500_gpio_platform_data *gpio; + struct ab8500_codec_platform_data *codec; }; extern int __devinit ab8500_init(struct ab8500 *ab8500, -- 1.7.8.3

3 2

[alsa-devel] Internal MIC is not working on Lenovo Y470
by Cyril Lashkevich 12 Jun '12

12 Jun '12

Hi, Internal MIC is not working on Lenovo Y470 with recent kernel (3.4.1) and alsa-lib (1.0.25). External mic connected to the mic jack works fine. alsa-info.sh output is in attachment. Thanks in advance, Cyril.

1 0

[alsa-devel] ALSA-SoC: Audio Peripheral without codec
by Prasant J 12 Jun '12

12 Jun '12

Hi, I'm working on linux-3.2.14 (TI-Beaglebone). After reading the kernel docs regarding the ALSA-SoC, what I found out is that ALSA-SoC framework also includes the Codec drivers. In my application, my processor want to communicate to another processor (TDM), but there is no Codec. How do I configure, ALSA-SoC, so that it ignores the Codec? Is there a dummy codec code already present for such a use case? Any suggestions? Regards, Prasant

1 0

[alsa-devel] [PATCH] ASoC: tegra+wm8903: turn of mic detect when card is removed
by Stephen Warren 12 Jun '12

12 Jun '12

From: Stephen Warren <swarren(a)nvidia.com> If mic detect is left enabled and the WM8903 detects a status change, the WM8903 driver will make a callback against the free()d jack, which will cause a crash. This problem can be triggered by fully initializing an audio card, then removing and re-inserting the machine driver module. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_wm8903.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 08b5fef..0c5bb33 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -213,6 +213,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_wm8903_remove(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + + wm8903_mic_detect(codec, NULL, 0, 0); + + return 0; +} + static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", @@ -233,6 +244,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .dai_link = &tegra_wm8903_dai, .num_links = 1, + .remove = tegra_wm8903_remove, + .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/2] dmaengine: Add wrapper for device_tx_status callback
by Lars-Peter Clausen 11 Jun '12

11 Jun '12

This patch adds a small inline wrapper for the devivce_tx_status callback of a dma device. This makes the source code of users of this function a bit more compact and a bit more legible. E.g.: -status = chan->device->device_tx_status(chan, cookie, &state) +status = dmaengine_tx_status(chan, cookie, &state) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/dmaengine.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 56bbc4d..1731628 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -650,6 +650,12 @@ static inline int dmaengine_resume(struct dma_chan *chan) return dmaengine_device_control(chan, DMA_RESUME, 0); } +static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *state) +{ + return chan->device->device_tx_status(chan, cookie, state); +} + static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc) { return desc->tx_submit(desc); -- 1.7.10

3 8

[alsa-devel] [PATCH 2/3] ASoC: Ux500: Add machine-driver
by Ola Lilja 11 Jun '12

11 Jun '12

Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 11 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/mop500.c | 115 +++++++++++ sound/soc/ux500/mop500_ab8500.c | 425 +++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.h | 22 ++ 5 files changed, 576 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 1d38515..069330d 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -19,3 +19,14 @@ config SND_SOC_UX500_PLAT_DMA select SND_SOC_DMAENGINE_PCM help Say Y if you want to enable the Ux500 platform-driver. + ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 4634bf0..cce0c11 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -5,3 +5,6 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 0000000..989114a --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,115 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + mop500_ab8500_remove(mop500_card); + + snd_soc_unregister_card(mop500_card); + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 0000000..2da8edb --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,425 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <mach/hardware.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Clocks */ +static const char * const enum_mclk[] = { + "SYSCLK", + "ULPCLK" +}; +enum mclk { + MCLK_SYSCLK, + MCLK_ULPCLK, +}; + +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk); + +/* Private data for machine-part MOP500<->AB8500 */ +struct mop500_ab8500_drvdata { + /* Clocks */ + enum mclk mclk_sel; + struct clk *clk_ptr_intclk; + struct clk *clk_ptr_sysclk; + struct clk *clk_ptr_ulpclk; +}; + +static inline const char *get_mclk_str(enum mclk mclk_sel) +{ + switch (mclk_sel) { + case MCLK_SYSCLK: + return "MCLK_SYSCLK"; + case MCLK_ULPCLK: + return "MCLK_ULPCLK"; + default: + return "Unknown"; + } +} + +static int mop500_ab8500_set_mclk(struct device *dev, + struct mop500_ab8500_drvdata *drvdata) +{ + int status; + struct clk *clk_ptr; + + if (IS_ERR(drvdata->clk_ptr_intclk)) { + dev_err(dev, + "%s: ERROR: intclk not initialized!\n", __func__); + return -EIO; + } + + switch (drvdata->mclk_sel) { + case MCLK_SYSCLK: + clk_ptr = drvdata->clk_ptr_sysclk; + break; + case MCLK_ULPCLK: + clk_ptr = drvdata->clk_ptr_ulpclk; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clk_ptr)) { + dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__, + get_mclk_str(drvdata->mclk_sel)); + return -EIO; + } + + status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr); + if (status) + dev_err(dev, + "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!", + __func__, get_mclk_str(drvdata->mclk_sel), status); + else + dev_dbg(dev, + "%s: intclk parent changed to %s.\n", + __func__, get_mclk_str(drvdata->mclk_sel)); + + return status; +} + +/* + * Control-events + */ + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + + ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; + + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + unsigned int val; + + val = (ucontrol->value.enumerated.item[0] != 0); + if (drvdata->mclk_sel == val) + return 0; + + drvdata->mclk_sel = val; + + return 1; +} + +/* + * Controls + */ + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_ENUM_EXT("Master Clock Select", + soc_enum_mclk, + mclk_input_control_get, mclk_input_control_put), + /* Digital interface - Clocks */ + SOC_SINGLE("Digital Interface Master Generator Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, + 1, 0), + SOC_SINGLE("Digital Interface 0 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, + 1, 0), + SOC_SINGLE("Digital Interface 1 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, + 1, 0), + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +int mop500_ab8500_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set audio-clock source */ + return mop500_ab8500_set_mclk(rtd->card->dev, + snd_soc_card_get_drvdata(rtd->card)); +} + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct device *dev = rtd->card->dev; + unsigned int fmt; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + driver_mode = (channels == 8) ? + DRIVERMODE_CODEC_ONLY : DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + sw_codec = (driver_mode == DRIVERMODE_NORMAL) ? sw_cpu : 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .startup = mop500_ab8500_startup, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_ab8500_drvdata *drvdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), + GFP_KERNEL); + snd_soc_card_set_drvdata(rtd->card, drvdata); + + /* Setup clocks */ + + drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk"); + if (IS_ERR(drvdata->clk_ptr_sysclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n", + __func__); + drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk"); + if (IS_ERR(drvdata->clk_ptr_ulpclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n", + __func__); + drvdata->clk_ptr_intclk = clk_get(dev, "intclk"); + if (IS_ERR(drvdata->clk_ptr_intclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n", + __func__); + + /* Set intclk default parent to ulpclk */ + drvdata->mclk_sel = MCLK_ULPCLK; + ret = mop500_ab8500_set_mclk(dev, drvdata); + if (ret < 0) + dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n", + __func__); + + drvdata->mclk_sel = MCLK_ULPCLK; + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + +void mop500_ab8500_remove(struct snd_soc_card *card) +{ + struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (drvdata->clk_ptr_sysclk != NULL) + clk_put(drvdata->clk_ptr_sysclk); + if (drvdata->clk_ptr_ulpclk != NULL) + clk_put(drvdata->clk_ptr_ulpclk); + if (drvdata->clk_ptr_intclk != NULL) + clk_put(drvdata->clk_ptr_intclk); + + snd_soc_card_set_drvdata(card, drvdata); +} diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 0000000..cca5b33 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,22 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); +void mop500_ab8500_remove(struct snd_soc_card *card); + +#endif -- 1.7.8.3

2 4

Re: [alsa-devel] [PATCH v4] ASoC: codecs: Add DA732x codec driver
by Mark Brown 11 Jun '12

11 Jun '12

On Mon, Jun 11, 2012 at 01:15:27PM +0100, Adam Thomson wrote: > This patch adds support for Dialog DA732x audio codecs. Applied, thanks.

1 0

[alsa-devel] [PATCH 1/8] ASoC: wm8996: Remove write sequencer registers from the defaults table
by Mark Brown 11 Jun '12

11 Jun '12

They aren't marked as readable and the feature is never used so they'll never get referenced. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 178 --------------------------------------------- 1 file changed, 178 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index efc4e9d..f24989f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -296,184 +296,6 @@ static struct reg_default wm8996_reg[] = { { WM8996_RIGHT_PDM_SPEAKER, 0x1 }, { WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 }, { WM8996_PDM_SPEAKER_VOLUME, 0x66 }, - { WM8996_WRITE_SEQUENCER_0, 0x1 }, - { WM8996_WRITE_SEQUENCER_1, 0x1 }, - { WM8996_WRITE_SEQUENCER_3, 0x6 }, - { WM8996_WRITE_SEQUENCER_4, 0x40 }, - { WM8996_WRITE_SEQUENCER_5, 0x1 }, - { WM8996_WRITE_SEQUENCER_6, 0xf }, - { WM8996_WRITE_SEQUENCER_7, 0x6 }, - { WM8996_WRITE_SEQUENCER_8, 0x1 }, - { WM8996_WRITE_SEQUENCER_9, 0x3 }, - { WM8996_WRITE_SEQUENCER_10, 0x104 }, - { WM8996_WRITE_SEQUENCER_12, 0x60 }, - { WM8996_WRITE_SEQUENCER_13, 0x11 }, - { WM8996_WRITE_SEQUENCER_14, 0x401 }, - { WM8996_WRITE_SEQUENCER_16, 0x50 }, - { WM8996_WRITE_SEQUENCER_17, 0x3 }, - { WM8996_WRITE_SEQUENCER_18, 0x100 }, - { WM8996_WRITE_SEQUENCER_20, 0x51 }, - { WM8996_WRITE_SEQUENCER_21, 0x3 }, - { WM8996_WRITE_SEQUENCER_22, 0x104 }, - { WM8996_WRITE_SEQUENCER_23, 0xa }, - { WM8996_WRITE_SEQUENCER_24, 0x60 }, - { WM8996_WRITE_SEQUENCER_25, 0x3b }, - { WM8996_WRITE_SEQUENCER_26, 0x502 }, - { WM8996_WRITE_SEQUENCER_27, 0x100 }, - { WM8996_WRITE_SEQUENCER_28, 0x2fff }, - { WM8996_WRITE_SEQUENCER_32, 0x2fff }, - { WM8996_WRITE_SEQUENCER_36, 0x2fff }, - { WM8996_WRITE_SEQUENCER_40, 0x2fff }, - { WM8996_WRITE_SEQUENCER_44, 0x2fff }, - { WM8996_WRITE_SEQUENCER_48, 0x2fff }, - { WM8996_WRITE_SEQUENCER_52, 0x2fff }, - { WM8996_WRITE_SEQUENCER_56, 0x2fff }, - { WM8996_WRITE_SEQUENCER_60, 0x2fff }, - { WM8996_WRITE_SEQUENCER_64, 0x1 }, - { WM8996_WRITE_SEQUENCER_65, 0x1 }, - { WM8996_WRITE_SEQUENCER_67, 0x6 }, - { WM8996_WRITE_SEQUENCER_68, 0x40 }, - { WM8996_WRITE_SEQUENCER_69, 0x1 }, - { WM8996_WRITE_SEQUENCER_70, 0xf }, - { WM8996_WRITE_SEQUENCER_71, 0x6 }, - { WM8996_WRITE_SEQUENCER_72, 0x1 }, - { WM8996_WRITE_SEQUENCER_73, 0x3 }, - { WM8996_WRITE_SEQUENCER_74, 0x104 }, - { WM8996_WRITE_SEQUENCER_76, 0x60 }, - { WM8996_WRITE_SEQUENCER_77, 0x11 }, - { WM8996_WRITE_SEQUENCER_78, 0x401 }, - { WM8996_WRITE_SEQUENCER_80, 0x50 }, - { WM8996_WRITE_SEQUENCER_81, 0x3 }, - { WM8996_WRITE_SEQUENCER_82, 0x100 }, - { WM8996_WRITE_SEQUENCER_84, 0x60 }, - { WM8996_WRITE_SEQUENCER_85, 0x3b }, - { WM8996_WRITE_SEQUENCER_86, 0x502 }, - { WM8996_WRITE_SEQUENCER_87, 0x100 }, - { WM8996_WRITE_SEQUENCER_88, 0x2fff }, - { WM8996_WRITE_SEQUENCER_92, 0x2fff }, - { WM8996_WRITE_SEQUENCER_96, 0x2fff }, - { WM8996_WRITE_SEQUENCER_100, 0x2fff }, - { WM8996_WRITE_SEQUENCER_104, 0x2fff }, - { WM8996_WRITE_SEQUENCER_108, 0x2fff }, - { WM8996_WRITE_SEQUENCER_112, 0x2fff }, - { WM8996_WRITE_SEQUENCER_116, 0x2fff }, - { WM8996_WRITE_SEQUENCER_120, 0x2fff }, - { WM8996_WRITE_SEQUENCER_124, 0x2fff }, - { WM8996_WRITE_SEQUENCER_128, 0x1 }, - { WM8996_WRITE_SEQUENCER_129, 0x1 }, - { WM8996_WRITE_SEQUENCER_131, 0x6 }, - { WM8996_WRITE_SEQUENCER_132, 0x40 }, - { WM8996_WRITE_SEQUENCER_133, 0x1 }, - { WM8996_WRITE_SEQUENCER_134, 0xf }, - { WM8996_WRITE_SEQUENCER_135, 0x6 }, - { WM8996_WRITE_SEQUENCER_136, 0x1 }, - { WM8996_WRITE_SEQUENCER_137, 0x3 }, - { WM8996_WRITE_SEQUENCER_138, 0x106 }, - { WM8996_WRITE_SEQUENCER_140, 0x61 }, - { WM8996_WRITE_SEQUENCER_141, 0x11 }, - { WM8996_WRITE_SEQUENCER_142, 0x401 }, - { WM8996_WRITE_SEQUENCER_144, 0x50 }, - { WM8996_WRITE_SEQUENCER_145, 0x3 }, - { WM8996_WRITE_SEQUENCER_146, 0x102 }, - { WM8996_WRITE_SEQUENCER_148, 0x51 }, - { WM8996_WRITE_SEQUENCER_149, 0x3 }, - { WM8996_WRITE_SEQUENCER_150, 0x106 }, - { WM8996_WRITE_SEQUENCER_151, 0xa }, - { WM8996_WRITE_SEQUENCER_152, 0x61 }, - { WM8996_WRITE_SEQUENCER_153, 0x3b }, - { WM8996_WRITE_SEQUENCER_154, 0x502 }, - { WM8996_WRITE_SEQUENCER_155, 0x100 }, - { WM8996_WRITE_SEQUENCER_156, 0x2fff }, - { WM8996_WRITE_SEQUENCER_160, 0x2fff }, - { WM8996_WRITE_SEQUENCER_164, 0x2fff }, - { WM8996_WRITE_SEQUENCER_168, 0x2fff }, - { WM8996_WRITE_SEQUENCER_172, 0x2fff }, - { WM8996_WRITE_SEQUENCER_176, 0x2fff }, - { WM8996_WRITE_SEQUENCER_180, 0x2fff }, - { WM8996_WRITE_SEQUENCER_184, 0x2fff }, - { WM8996_WRITE_SEQUENCER_188, 0x2fff }, - { WM8996_WRITE_SEQUENCER_192, 0x1 }, - { WM8996_WRITE_SEQUENCER_193, 0x1 }, - { WM8996_WRITE_SEQUENCER_195, 0x6 }, - { WM8996_WRITE_SEQUENCER_196, 0x40 }, - { WM8996_WRITE_SEQUENCER_197, 0x1 }, - { WM8996_WRITE_SEQUENCER_198, 0xf }, - { WM8996_WRITE_SEQUENCER_199, 0x6 }, - { WM8996_WRITE_SEQUENCER_200, 0x1 }, - { WM8996_WRITE_SEQUENCER_201, 0x3 }, - { WM8996_WRITE_SEQUENCER_202, 0x106 }, - { WM8996_WRITE_SEQUENCER_204, 0x61 }, - { WM8996_WRITE_SEQUENCER_205, 0x11 }, - { WM8996_WRITE_SEQUENCER_206, 0x401 }, - { WM8996_WRITE_SEQUENCER_208, 0x50 }, - { WM8996_WRITE_SEQUENCER_209, 0x3 }, - { WM8996_WRITE_SEQUENCER_210, 0x102 }, - { WM8996_WRITE_SEQUENCER_212, 0x61 }, - { WM8996_WRITE_SEQUENCER_213, 0x3b }, - { WM8996_WRITE_SEQUENCER_214, 0x502 }, - { WM8996_WRITE_SEQUENCER_215, 0x100 }, - { WM8996_WRITE_SEQUENCER_216, 0x2fff }, - { WM8996_WRITE_SEQUENCER_220, 0x2fff }, - { WM8996_WRITE_SEQUENCER_224, 0x2fff }, - { WM8996_WRITE_SEQUENCER_228, 0x2fff }, - { WM8996_WRITE_SEQUENCER_232, 0x2fff }, - { WM8996_WRITE_SEQUENCER_236, 0x2fff }, - { WM8996_WRITE_SEQUENCER_240, 0x2fff }, - { WM8996_WRITE_SEQUENCER_244, 0x2fff }, - { WM8996_WRITE_SEQUENCER_248, 0x2fff }, - { WM8996_WRITE_SEQUENCER_252, 0x2fff }, - { WM8996_WRITE_SEQUENCER_256, 0x60 }, - { WM8996_WRITE_SEQUENCER_258, 0x601 }, - { WM8996_WRITE_SEQUENCER_260, 0x50 }, - { WM8996_WRITE_SEQUENCER_262, 0x100 }, - { WM8996_WRITE_SEQUENCER_264, 0x1 }, - { WM8996_WRITE_SEQUENCER_266, 0x104 }, - { WM8996_WRITE_SEQUENCER_267, 0x100 }, - { WM8996_WRITE_SEQUENCER_268, 0x2fff }, - { WM8996_WRITE_SEQUENCER_272, 0x2fff }, - { WM8996_WRITE_SEQUENCER_276, 0x2fff }, - { WM8996_WRITE_SEQUENCER_280, 0x2fff }, - { WM8996_WRITE_SEQUENCER_284, 0x2fff }, - { WM8996_WRITE_SEQUENCER_288, 0x2fff }, - { WM8996_WRITE_SEQUENCER_292, 0x2fff }, - { WM8996_WRITE_SEQUENCER_296, 0x2fff }, - { WM8996_WRITE_SEQUENCER_300, 0x2fff }, - { WM8996_WRITE_SEQUENCER_304, 0x2fff }, - { WM8996_WRITE_SEQUENCER_308, 0x2fff }, - { WM8996_WRITE_SEQUENCER_312, 0x2fff }, - { WM8996_WRITE_SEQUENCER_316, 0x2fff }, - { WM8996_WRITE_SEQUENCER_320, 0x61 }, - { WM8996_WRITE_SEQUENCER_322, 0x601 }, - { WM8996_WRITE_SEQUENCER_324, 0x50 }, - { WM8996_WRITE_SEQUENCER_326, 0x102 }, - { WM8996_WRITE_SEQUENCER_328, 0x1 }, - { WM8996_WRITE_SEQUENCER_330, 0x106 }, - { WM8996_WRITE_SEQUENCER_331, 0x100 }, - { WM8996_WRITE_SEQUENCER_332, 0x2fff }, - { WM8996_WRITE_SEQUENCER_336, 0x2fff }, - { WM8996_WRITE_SEQUENCER_340, 0x2fff }, - { WM8996_WRITE_SEQUENCER_344, 0x2fff }, - { WM8996_WRITE_SEQUENCER_348, 0x2fff }, - { WM8996_WRITE_SEQUENCER_352, 0x2fff }, - { WM8996_WRITE_SEQUENCER_356, 0x2fff }, - { WM8996_WRITE_SEQUENCER_360, 0x2fff }, - { WM8996_WRITE_SEQUENCER_364, 0x2fff }, - { WM8996_WRITE_SEQUENCER_368, 0x2fff }, - { WM8996_WRITE_SEQUENCER_372, 0x2fff }, - { WM8996_WRITE_SEQUENCER_376, 0x2fff }, - { WM8996_WRITE_SEQUENCER_380, 0x2fff }, - { WM8996_WRITE_SEQUENCER_384, 0x60 }, - { WM8996_WRITE_SEQUENCER_386, 0x601 }, - { WM8996_WRITE_SEQUENCER_388, 0x61 }, - { WM8996_WRITE_SEQUENCER_390, 0x601 }, - { WM8996_WRITE_SEQUENCER_392, 0x50 }, - { WM8996_WRITE_SEQUENCER_394, 0x300 }, - { WM8996_WRITE_SEQUENCER_396, 0x1 }, - { WM8996_WRITE_SEQUENCER_398, 0x304 }, - { WM8996_WRITE_SEQUENCER_400, 0x40 }, - { WM8996_WRITE_SEQUENCER_402, 0xf }, - { WM8996_WRITE_SEQUENCER_404, 0x1 }, - { WM8996_WRITE_SEQUENCER_407, 0x100 }, }; static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); -- 1.7.10

1 7

Re: [alsa-devel] [PATCH] ASoC: codecs: Add AB8500 codec-driver
by Mark Brown 11 Jun '12

11 Jun '12

On Mon, Jun 11, 2012 at 01:06:24PM +0200, Ola Lilja wrote: > Add codec-driver for ST-Ericsson AB8500 mixed-signal ASIC. > > Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> I said I'd already applied the patch...

1 0

[alsa-devel] [RFC] usb-audio: Use a table of mixer controls
by Mark Hills 11 Jun '12

11 Jun '12

Allow mixer controls to be provided clearly in a table, to avoid quantity of error checking at each use. This is definitely clearer than the current code. But going forward, how should this work with the tables defined in mixer_maps.c? I've made this generic and tried to follow existing conventions for similar tables (eg. use a terminator, not array size), comments welcome. --- sound/usb/mixer_quirks.c | 104 +++++++++++++++++++++------------------------- 1 files changed, 48 insertions(+), 56 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 41f4b69..b80b7e7 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -42,6 +42,13 @@ extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; +struct std_mono_table { + unsigned int unitid, control, cmask; + int val_type; + const char *name; + snd_kcontrol_tlv_rw_t *tlv_callback; +}; + /* private_free callback */ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -114,6 +121,24 @@ static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, } /* + * Create a set of standard UAC controls from a table + */ +static int snd_create_std_mono_table(struct usb_mixer_interface *mixer, + struct std_mono_table *t) +{ + int err; + + while (t->name != NULL) { + err = snd_create_std_mono_ctl(mixer, t->unitid, t->control, + t->cmask, t->val_type, t->name, t->tlv_callback); + if (err < 0) + return err; + } + + return 0; +} + +/* * Sound Blaster remote control configuration * * format of remote control data: @@ -916,61 +941,6 @@ static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer) return 0; } - -/* - * Create mixer for Electrix Ebox-44 - * - * The mixer units from this device are corrupt, and even where they - * are valid they presents mono controls as L and R channels of - * stereo. So we create a good mixer in code. - */ - -static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer) -{ - int err; - - err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, - "Headphone Playback Switch", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16, - "Headphone A Mix Playback Volume", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16, - "Headphone B Mix Playback Volume", NULL); - if (err < 0) - return err; - - err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, - "Output Playback Switch", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16, - "Output A Playback Volume", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16, - "Output B Playback Volume", NULL); - if (err < 0) - return err; - - err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, - "Input Capture Switch", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16, - "Input A Capture Volume", NULL); - if (err < 0) - return err; - err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16, - "Input B Capture Volume", NULL); - if (err < 0) - return err; - - return 0; -} - void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, unsigned char samplerate_id) { @@ -990,6 +960,27 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * The mixer units for Ebox-44 are corrupt, and even where they + * are valid they presents mono controls as L and R channels of + * stereo. So we create provide a good mixer here. + */ +struct std_mono_table ebox44_table[] = { + { 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Headphone Playback Switch", NULL }, + { 4, 2, 0x1, USB_MIXER_S16, "Headphone A Mix Playback Volume", NULL }, + { 4, 2, 0x2, USB_MIXER_S16, "Headphone B Mix Playback Volume", NULL }, + + { 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Output Playback Switch", NULL }, + { 7, 2, 0x1, USB_MIXER_S16, "Output A Playback Volume", NULL }, + { 7, 2, 0x2, USB_MIXER_S16, "Output B Playback Volume", NULL }, + + { 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Input Capture Switch", NULL }, + { 10, 2, 0x1, USB_MIXER_S16, "Input A Capture Volume", NULL }, + { 10, 2, 0x2, USB_MIXER_S16, "Input B Capture Volume", NULL }, + + { } +}; + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1035,7 +1026,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */ - err = snd_ebox44_create_mixer(mixer); + /* detection is disabled in mixer_maps.c */ + err = snd_create_std_mono_table(mixer, ebox44_table); break; } -- 1.7.4.4

2 7

[alsa-devel] [PATCH 1/4] utils/alsa-info.sh: adding system firmware version to the defaults collected under "DMI Information"
by David Henningsson 11 Jun '12

11 Jun '12

From: Chris Van Hoof <vanhoof(a)canonical.com> Signed-off-by: Chris Van Hoof <vanhoof(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- utils/alsa-info.sh | 3 +++ 1 file changed, 3 insertions(+) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index fd7df96..46d2ebb 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -424,10 +424,12 @@ if [ -d /sys/class/dmi/id ]; then DMI_SYSTEM_MANUFACTURER=$(cat /sys/class/dmi/id/sys_vendor 2>/dev/null) DMI_SYSTEM_PRODUCT_NAME=$(cat /sys/class/dmi/id/product_name 2>/dev/null) DMI_SYSTEM_PRODUCT_VERSION=$(cat /sys/class/dmi/id/product_version 2>/dev/null) + DMI_SYSTEM_FIRMWARE_VERSION=$(cat /sys/class/dmi/id/bios_version 2>/dev/null) elif [ -x $DMIDECODE ]; then DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) DMI_SYSTEM_PRODUCT_VERSION=$($DMIDECODE -s system-version 2>/dev/null) + DMI_SYSTEM_FIRMWARE_VERSION=$($DMIDECODE -s bios-version 2>/dev/null) fi cat /proc/asound/modules 2>/dev/null|awk {'print $2'}>$TEMPDIR/alsamodules.tmp @@ -469,6 +471,7 @@ echo "" >> $FILE echo "Manufacturer: $DMI_SYSTEM_MANUFACTURER" >> $FILE echo "Product Name: $DMI_SYSTEM_PRODUCT_NAME" >> $FILE echo "Product Version: $DMI_SYSTEM_PRODUCT_VERSION" >> $FILE +echo "Firmware Version: $DMI_SYSTEM_FIRMWARE_VERSION" >> $FILE echo "" >> $FILE echo "" >> $FILE echo "!!Kernel Information" >> $FILE -- 1.7.9.5

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Re: [alsa-devel] [PATCH v3] ASoC: codecs: Add DA732x codec driver
by Adam Thomson 11 Jun '12

11 Jun '12

On Wed, Jun 11, 2012 at 10:02AM, Mark Brown wrote: > > ASoC should be the same in both, my for-next is what turns up in -next. > The change would get applied on for-3.6. Ok, thanks for the info. Will base against 'for-3.6'.

1 0

[alsa-devel] [PATCH] ASOC: utils: Fix the dummy_codec
by Selma Bensaid 11 Jun '12

11 Jun '12

This patch consists in settings the channels_min, rates and formats of the dummy codec DAI driver to enable the creation of the PCM streams using it. Signed-off-by: Selma Bensaid <selma.bensaid(a)intel.com> --- sound/soc/soc-utils.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 6005370..465a094 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -89,9 +89,27 @@ static struct snd_soc_platform_driver dummy_platform = { .ops = &dummy_dma_ops, }; +#define NULL_FORMATS \ + (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 |\ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 |\ + SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32) + static struct snd_soc_codec_driver dummy_codec; static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", + .playback = { + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = NULL_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = NULL_FORMATS, + }, + .ops = NULL, }; static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) -- 1.7.0.4 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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Re: [alsa-devel] [PATCH v3] ASoC: codecs: Add DA732x codec driver
by Adam Thomson 11 Jun '12

11 Jun '12

On Wed, Jun 11, 2012 at 05:41AM, Mark Brown wrote: > This doesn't apply, please always submit new code against current > development code for the subsystem. You've also got something else > wrong with your setup, the Date: header for your mail is "Date: $(date > -R)" which is upsetting the tools. Apologies. Was using linux-next as my base code. Will re-send changes against your ASoC repository. Assume it should be against the 'for-next' head? > Please also send followups adding the specific devices to the ID table. Currently there's just one device ID. Will add that now, and when new devices are available then will then update accordingly.

2 1

Re: [alsa-devel] [PATCH 1/3] ASoC: codecs: Add AB8500 codec-driver
by Mark Brown 11 Jun '12

11 Jun '12

On Thu, Jun 07, 2012 at 02:00:21PM +0200, Ola Lilja wrote: > Add codec-driver for ST-Ericsson AB8500 mixed-signal ASIC. So, I applied this even though you left the API for doing platform data in the machine driver in place since I see further patches which should allow these to be unexported. It'd have been better if you'd removed them from this patch as requested, that way history would be a bit cleaner and I'd not have to decide if the current stuff is OK.

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[alsa-devel] [PATCH 1/5] ASoC: wm8904: Fix GPIO and MICBIAS initialisation for regmap conversion
by Mark Brown 11 Jun '12

11 Jun '12

We no longer have a flat ASoC cache so can't peer directly into the array any more but should instead use the register I/O functions to update the cache. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 560a9a47..461c58a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2084,7 +2084,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); struct wm8904_pdata *pdata = wm8904->pdata; - u16 *reg_cache = codec->reg_cache; int ret, i; codec->cache_sync = 1; @@ -2180,14 +2179,18 @@ static int wm8904_probe(struct snd_soc_codec *codec) if (!pdata->gpio_cfg[i]) continue; - reg_cache[WM8904_GPIO_CONTROL_1 + i] - = pdata->gpio_cfg[i] & 0xffff; + regmap_update_bits(wm8904->regmap, + WM8904_GPIO_CONTROL_1 + i, + 0xffff, + pdata->gpio_cfg[i]); } /* Zero is the default value for these anyway */ for (i = 0; i < WM8904_MIC_REGS; i++) - reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i] - = pdata->mic_cfg[i]; + regmap_update_bits(wm8904->regmap, + WM8904_MIC_BIAS_CONTROL_0 + i, + 0xffff, + pdata->mic_cfg[i]); } /* Set Class W by default - this will be managed by the Class -- 1.7.10

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[alsa-devel] [PATCH] ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word
by Ricardo Neri 11 Jun '12

11 Jun '12

Hi Mark, Liam, This patch is to fix an bug that I found in K3.5-rc1. As the IEC-60958 channel status word is not initialized in hw_params, the params from previous calls corrupt those of the current playback. BR Ricardo Ricardo Neri (1): ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word sound/soc/omap/omap-hdmi.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) -- 1.7.5.4

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Re: [alsa-devel] [PATCH v3] ASoC: codecs: Add DA732x codec driver
by Mark Brown 11 Jun '12

11 Jun '12

On Wed, Jun 06, 2012 at 04:35:39PM +0000, Adam Thomson wrote: > This patch adds support for the Dialog DA732x audio codec. > > Signed-off-by: Michal Hajduk <Michal.Hajduk(a)diasemi.com> > Signed-off-by: Adam Thomson <Adam.Thomson(a)diasemi.com> This doesn't apply, please always submit new code against current development code for the subsystem. You've also got something else wrong with your setup, the Date: header for your mail is "Date: $(date -R)" which is upsetting the tools. Please also send followups adding the specific devices to the ID table.

1 0

Re: [alsa-devel] [PATCH 2/2 V3] ASoC: fix pxa-ssp compiling issue under mach-mmp
by Mark Brown 11 Jun '12

11 Jun '12

On Mon, Jun 11, 2012 at 09:52:36AM +0800, Qiao Zhou wrote: > On 06/04/2012 10:41 AM, Qiao Zhou wrote: Delete irrelevant quotes from your mails! There's about 180 lines of text here and only one line of new text buried right at the bttom. > > pxa-ssp.c uses API like cpu_is_pxa3xx(), cpu_is_pxa2xx(), which is > > defined under arch-pxa architecture, and drivers under mach-mmp > do you have any suggestion/comment about the patch series? I'm hoping that the arch/arm patch can can get reviewed... -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ASoC: lm49453: Remove a unneeded version.h include
by Jesper Juhl 11 Jun '12

11 Jun '12

Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/lm49453.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f1..c1bc945 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -12,7 +12,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.7.10.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

2 1

[alsa-devel] [PATCH] ASoC: cs42l52: Remove unneeded version.h include
by Jesper Juhl 11 Jun '12

11 Jun '12

Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/cs42l52.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index a710941..ec03abc 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -14,7 +14,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.7.10.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

2 1

[PATCH] USB: fix gathering of interface associations
by Daniel Mack 10 Jun '12

10 Jun '12

TEAC's UD-H01 (and probably other devices) have a gap in the interface number allocation in their descriptors: Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 220 bNumInterfaces 3 [...] Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 [...] Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 [...] Once a configuration is selected on this device, usb_set_configuration() walks the known interfaces of a given configuration and calls find_iad() on each of them to set the interface association pointer the interface is included in. The problem here is that the loop varibale index is taken for the interface number in the comparison logic that gathers the association. Which is fine as long as the descriptors are sane. In the case above, however, the logic gets out of sync and the interface association fields of all interfaces beyond the interface number gap are wrong. Fix this by passing the interface's bInterfaceNumber to find_iad() instead. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: bEN <ml_all(a)circa.be> Reported-by: Ivan Perrone <ivanperrone(a)hotmail.com> Tested-by: Sergio Tabanelli <sergio.tabanelli(a)gmail.com> Cc: <stable(a)kernel.org> --- drivers/usb/core/message.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/usb/core/message.c b/drivers/usb/core/message.c index b548cf1..bdd1c67 100644 --- a/drivers/usb/core/message.c +++ b/drivers/usb/core/message.c @@ -1838,7 +1838,6 @@ free_interfaces: intfc = cp->intf_cache[i]; intf->altsetting = intfc->altsetting; intf->num_altsetting = intfc->num_altsetting; - intf->intf_assoc = find_iad(dev, cp, i); kref_get(&intfc->ref); alt = usb_altnum_to_altsetting(intf, 0); @@ -1851,6 +1850,8 @@ free_interfaces: if (!alt) alt = &intf->altsetting[0]; + intf->intf_assoc = + find_iad(dev, cp, alt->desc.bInterfaceNumber); intf->cur_altsetting = alt; usb_enable_interface(dev, intf, true); intf->dev.parent = &dev->dev; -- 1.7.10.2 --------------050305000101090703040603--

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[alsa-devel] Possible option for driver development for Creamware/Sonic Core soon
by Dan Swain 10 Jun '12

10 Jun '12

Dear all, I've recently acquired one of the less-recent Creamware Luna cards. After reading on Nabble that someone expressed an interest in ALSA drivers for these cards two years ago (and hearing from Paul Davis that he'd been unsuccessful in gaining enthusiasm from Creamware), I decided to get in touch with the community that supports these cards on forums.planetz.com It appears that in 2006, Sonic Core bought Creamware, and began developing their own framework to program the DSPs on the original cards, plus some new creations by them. This platform is known as Scope, and the most current version is 5.1, which is compatible with Windows 32 and 64-bit operating systems. However, they (Sonic Core) are working on Scope 6.0, which is intended to be a cross-platform system. From what I've been reading, they are using the JUCE framework as the basis of their development for this version, so they can target (primarily) Windows AND Mac OSX systems. Some interest has been noted for a Linux version too, after I started this thread: http://forums.planetz.com/viewtopic.php?f=5&t=31065 _From what I'm aware, the Sonic Core developers drop in to that forum now and again, so it may be possible to get informed responses. The plug-in base for these cards is quite extensive, even with the free ones developed by community members, particularly synthesisers, so it would make a wonderful addition to the Linux audio world I'm new to Sonic Core as well, and unfortunately I'm not a programmer, though I wouldn't be averse to trying to help in any way I can. My understanding here is that all that needs to be developed is a driver that connects to Sonic Core's framework. Thanks for taking the time to read, and I hope this sparks some interest with the ALSA community! Kind Regards, Dan -- Dan Swain <sp4md4n(a)gmail.com>

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Re: [alsa-devel] Help requested: new HSS1394 MIDI back-end
by Jonathan Woithe 09 Jun '12

09 Jun '12

Clemens Ladisch wrote: > Sean M. Pappalardo - D.J. Pegasus wrote: > > As soon as I start the JACK server to handle the audio interface on > > the SCS.1m, the ALSA driver stops sending MIDI for that device unless > > I stop JACK and power-cycle the device. Its last words are: > > F0 00 01 60 48 53 53 0F 03 00 03 00 00 0B 01 F7 > > This is a ping response (tag F3). Apparently, Jack's FFADO driver > reinitializes the device and registers its own address for received > packets. Since it does _not_ handle MIDI messages, it really should > not do this. If we're talking about the ARM handler for HSS1394 messages, then I suspect the reason it sets this up is that Pieter initially fully intended for FFADO to pass these messages back. Obviously if this were to be done then an ARM handler would be necessary, but that's not how things are panning out. To prevent the ARM handler being registered by FFADO it should only be ncessary to comment out the initMessageHandler() call in the ScsDevice::discover() method. I'm happy to make this change since it seems to make sense to do so given these latest developments. A preliminary approach is in FFADO svn revision 2163 if you want to check it out. Note that this will break FFADO's test-scs program - a detail I'll tidy up if the overall approach seems to work. The other related issue is whether the other device setup done by FFADO interferes with the MIDI stream transmission. I guess that's most easily tested by trying FFADO rev 2163 and seeing if there are other issues remaining. Regards jonathan

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Re: [alsa-devel] [PATCH 0/2] HD-audio HDMI regression fixes with VGA-switcheroo
by Takashi Iwai 09 Jun '12

09 Jun '12

At Fri, 08 Jun 2012 22:17:29 +0200, Jörg-Volker Peetz wrote: > > Takashi Iwai wrote, on 06/08/12 17:52: > > At Fri, 08 Jun 2012 17:45:17 +0200, > > Jörg-Volker Peetz wrote: > >> > >> Hello Takashi, > >> > >> Takashi Iwai wrote, on 06/08/12 15:03: > >>> At Fri, 08 Jun 2012 13:26:57 +0200, > >>> Jörg-Volker Peetz wrote: > >>>> > >>>> Takashi Iwai wrote, on 06/07/12 12:15: > >>>>> Hi, > >>>>> > >>>>> this is a series of patches to fix the regressions of HD-audio HDMI > >>>>> on D-GPUs in 3.5-rc1 due to the support of VGA-switcheroo audio clients. > >>>>> > >>>>> The first patch adds a new helper function to vga-switcheroo and the > >>>>> second just uses that instead of an open code. > >>>>> > >>>>> Dave, if the first patch is OK, I'm going to apply it though sound tree. > >>>>> Let me know if any problem is found. > >>>>> > >>>>> Joerg, could you check whether this doesn't break your setup, too? > >>>>> > >>>>> > >>>>> thanks, > >>>>> > >>>>> Takashi > >>>> > >>>> Hello Takashi, > >>>> > >>>> I applied both your patches on 3.5-rc1. With this kernel I made three tests. > >>>> 1) After booting I switched off the discrete GPU via vga_switcheroo. > >>>> Then ca. 210 times the message > >>>> > >>>> hda-intel: spurious response 0x0:0x0, last cmd=0x170503 > >>>> > >>>> appears on the console and in the logs. > >>>> Starting X and using the built-in USB web-cam work, as well as sound with > >>>> mplayer2 works. > >>>> > >>>> 2) After booting I switched to the discrete GPU via > >>>> > >>>> echo -n DDIS > /sys/kernel/debugfs/vgaswitcheroo/switch > >>>> > >>>> Then I started X via startx. The desktop screen appears but the system > >>>> freezes. No reaction on keyboard or touchpad input. The computer is not > >>>> reachable via ethernet wire. > >>> > >>> Hm, these issues aren't seen on 3.5-rc1 before my patches? > >>> If so, we need to fix now. > >>> > >> No, these issues aren't seen on 3.5-rc1 without your patches. I'm able to start > >> X with the discrete GPU in use, direct rendering and the built-in USB camera work. > > > > OK, it means that the switching from the audio-ON to audio-OFF doesn't > > work. Could you try the patch below in additon? > > > > Applying all your three patches on 3.5-rc1 makes the discrete GPU work on my > system: after mounting debugfs and switching to "DDIS", I can start X and direct > rendering, sound and built-in USB camera work. Good to hear. Dave, is it OK to apply the patch below through sound tree? thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] vga_switcheroo: Enable/disable audio clients at the right time The audio clients have to be disabled before disabling the VGA and switching. Similarly, enabling the audio client should be done at last. Otherwise the audio-side operation stalls, eventually leading to Oops or lockups. Tested-by: Jörg-Volker Peetz <jvpeetz(a)web.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index 38f9534..e24ad99 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -291,8 +291,6 @@ static int vga_switchto_stage1(struct vga_switcheroo_client *new_client) vga_switchon(new_client); vga_set_default_device(new_client->pdev); - set_audio_state(new_client->id, VGA_SWITCHEROO_ON); - return 0; } @@ -308,6 +306,8 @@ static int vga_switchto_stage2(struct vga_switcheroo_client *new_client) active->active = false; + set_audio_state(active->id, VGA_SWITCHEROO_OFF); + if (new_client->fb_info) { struct fb_event event; event.info = new_client->fb_info; @@ -321,11 +321,11 @@ static int vga_switchto_stage2(struct vga_switcheroo_client *new_client) if (new_client->ops->reprobe) new_client->ops->reprobe(new_client->pdev); - set_audio_state(active->id, VGA_SWITCHEROO_OFF); - if (active->pwr_state == VGA_SWITCHEROO_ON) vga_switchoff(active); + set_audio_state(new_client->id, VGA_SWITCHEROO_ON); + new_client->active = true; return 0; } @@ -371,8 +371,9 @@ vga_switcheroo_debugfs_write(struct file *filp, const char __user *ubuf, /* pwr off the device not in use */ if (strncmp(usercmd, "OFF", 3) == 0) { list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->active) + if (client->active || client_is_audio(client)) continue; + set_audio_state(client->id, VGA_SWITCHEROO_OFF); if (client->pwr_state == VGA_SWITCHEROO_ON) vga_switchoff(client); } @@ -381,10 +382,11 @@ vga_switcheroo_debugfs_write(struct file *filp, const char __user *ubuf, /* pwr on the device not in use */ if (strncmp(usercmd, "ON", 2) == 0) { list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->active) + if (client->active || client_is_audio(client)) continue; if (client->pwr_state == VGA_SWITCHEROO_OFF) vga_switchon(client); + set_audio_state(client->id, VGA_SWITCHEROO_ON); } goto out; } -- 1.7.10.4

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Re: [alsa-devel] [Bisected] ALSA: snd-usb: New streaming logic causes distortions on Burr-Brown DAC
by Takashi Iwai 09 Jun '12

09 Jun '12

At Fri, 8 Jun 2012 20:11:54 +0200, Markus Trippelsdorf wrote: > > On 2012.06.08 at 17:50 +0000, Jamie Heilman wrote: > > Markus Trippelsdorf wrote: > > > I hear distortions on my Burr-Brown USB DAC whenever the (software) > > > volume is set to some value lower than 100% in mpd. The lower the volume > > > the bigger are the resulting distortions. > > > > see patch: http://marc.info/?l=linux-kernel&m=133905533312625&w=2 > > > > You're probably getting distortion because mpd decided it would use > > a stream format that's only 8bit. > > Yes. The patch fixes the issue. Many thanks. The patch is already in sound git tree, but missed in rc2. It should be in rc3. thanks, Takashi

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[alsa-devel] [PATCH] ASoC: wm8903: disable mic detect irqs until jack registered
by Stephen Warren 09 Jun '12

09 Jun '12

From: Stephen Warren <swarren(a)nvidia.com> If either mic detection interrupt is unmasked when the IRQ handler is first registered, they might fire immediately. However, at this point, wm8903->mic_jack cannot have been set, since wm8903_mic_detect() can only operate after wm8903_probe() has completed. If the IRQ fires at this point, wm8903_irq() will use the uninitialized or stale mic_jack, and crash. This problem can be triggered by fully initializing an audio card, then removing and re-inserting the machine driver module. This would leave mic detection enabled in HW, and mic_jack set to a stale value. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This is useful in 3.5. As an aside, I wondered instead about modifying wm8903_remove() to disable mic detection, and clear any status. However, even if we do that, I think we still want to make this change too, since I think we want probe() to work the first time without relying on previous HW state? sound/soc/codecs/wm8903.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 304b5cf..d00a2b3 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1946,7 +1946,16 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL, WM8903_IRQ_POL, irq_pol); - + + /* + * Ensure mic detection interrupts are masked until after + * wm8903_mic_detect() is called, so that wm8903_irq() has + * a valid wm8903->mic_jack() to report on. + */ + snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, + WM8903_MICDET_EINT | WM8903_MICSHRT_EINT); + ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq, trigger | IRQF_ONESHOT, "wm8903", codec); -- 1.7.0.4

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[alsa-devel] [PATCH 1/2] ASoC: wm8904: Convert to module_i2c_driver()
by Mark Brown 09 Jun '12

09 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index db94d10..02bc2ca 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2313,23 +2313,7 @@ static struct i2c_driver wm8904_i2c_driver = { .id_table = wm8904_i2c_id, }; -static int __init wm8904_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8904_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8904_modinit); - -static void __exit wm8904_exit(void) -{ - i2c_del_driver(&wm8904_i2c_driver); -} -module_exit(wm8904_exit); +module_i2c_driver(wm8904_i2c_driver); MODULE_DESCRIPTION("ASoC WM8904 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -- 1.7.10

1 1

Re: [alsa-devel] [PATCH 0/2] HD-audio HDMI regression fixes with VGA-switcheroo
by Takashi Iwai 08 Jun '12

08 Jun '12

At Fri, 08 Jun 2012 17:45:17 +0200, Jörg-Volker Peetz wrote: > > Hello Takashi, > > Takashi Iwai wrote, on 06/08/12 15:03: > > At Fri, 08 Jun 2012 13:26:57 +0200, > > Jörg-Volker Peetz wrote: > >> > >> Takashi Iwai wrote, on 06/07/12 12:15: > >>> Hi, > >>> > >>> this is a series of patches to fix the regressions of HD-audio HDMI > >>> on D-GPUs in 3.5-rc1 due to the support of VGA-switcheroo audio clients. > >>> > >>> The first patch adds a new helper function to vga-switcheroo and the > >>> second just uses that instead of an open code. > >>> > >>> Dave, if the first patch is OK, I'm going to apply it though sound tree. > >>> Let me know if any problem is found. > >>> > >>> Joerg, could you check whether this doesn't break your setup, too? > >>> > >>> > >>> thanks, > >>> > >>> Takashi > >> > >> Hello Takashi, > >> > >> I applied both your patches on 3.5-rc1. With this kernel I made three tests. > >> 1) After booting I switched off the discrete GPU via vga_switcheroo. > >> Then ca. 210 times the message > >> > >> hda-intel: spurious response 0x0:0x0, last cmd=0x170503 > >> > >> appears on the console and in the logs. > >> Starting X and using the built-in USB web-cam work, as well as sound with > >> mplayer2 works. > >> > >> 2) After booting I switched to the discrete GPU via > >> > >> echo -n DDIS > /sys/kernel/debugfs/vgaswitcheroo/switch > >> > >> Then I started X via startx. The desktop screen appears but the system > >> freezes. No reaction on keyboard or touchpad input. The computer is not > >> reachable via ethernet wire. > > > > Hm, these issues aren't seen on 3.5-rc1 before my patches? > > If so, we need to fix now. > > > No, these issues aren't seen on 3.5-rc1 without your patches. I'm able to start > X with the discrete GPU in use, direct rendering and the built-in USB camera work. OK, it means that the switching from the audio-ON to audio-OFF doesn't work. Could you try the patch below in additon? > Probably unrelated: in both cases I'm unable to lower the screen brightness from > highest value. Without your patches it appears a warning: > > ACPI Warning: _BQC returned an invalid level (20120320/video-472) Hmm, I see no relation between this and the HDMI audio part. It's of course possible that some non-working state gives side-effects... thanks, Takashi --- diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index eb4f64f..5b3c7d1 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -304,8 +304,6 @@ static int vga_switchto_stage1(struct vga_switcheroo_client *new_client) vga_switchon(new_client); vga_set_default_device(new_client->pdev); - set_audio_state(new_client->id, VGA_SWITCHEROO_ON); - return 0; } @@ -321,6 +319,8 @@ static int vga_switchto_stage2(struct vga_switcheroo_client *new_client) active->active = false; + set_audio_state(active->id, VGA_SWITCHEROO_OFF); + if (new_client->fb_info) { struct fb_event event; event.info = new_client->fb_info; @@ -334,11 +334,11 @@ static int vga_switchto_stage2(struct vga_switcheroo_client *new_client) if (new_client->ops->reprobe) new_client->ops->reprobe(new_client->pdev); - set_audio_state(active->id, VGA_SWITCHEROO_OFF); - if (active->pwr_state == VGA_SWITCHEROO_ON) vga_switchoff(active); + set_audio_state(new_client->id, VGA_SWITCHEROO_ON); + new_client->active = true; return 0; } @@ -384,8 +384,9 @@ vga_switcheroo_debugfs_write(struct file *filp, const char __user *ubuf, /* pwr off the device not in use */ if (strncmp(usercmd, "OFF", 3) == 0) { list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->active) + if (client->active || client_is_audio(client)) continue; + set_audio_state(client->id, VGA_SWITCHEROO_OFF); if (client->pwr_state == VGA_SWITCHEROO_ON) vga_switchoff(client); } @@ -394,10 +395,11 @@ vga_switcheroo_debugfs_write(struct file *filp, const char __user *ubuf, /* pwr on the device not in use */ if (strncmp(usercmd, "ON", 2) == 0) { list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->active) + if (client->active || client_is_audio(client)) continue; if (client->pwr_state == VGA_SWITCHEROO_OFF) vga_switchon(client); + set_audio_state(client->id, VGA_SWITCHEROO_ON); } goto out; }

1 0

Re: [alsa-devel] [PATCH v1 3/5] ASOC: mmp: add sspa support
by Mark Brown 08 Jun '12

08 Jun '12

On Mon, Jun 04, 2012 at 02:37:29PM +0800, Zhangfei Gao wrote: > The SSPA is a configurable multi-channel audio serial (TDM) interface. > It's configurable at runtime to support up to 128 channels and the > number of bits per sample: 8, 12, 16, 20, 24 and 32 bits. It also > support stereo format: I2S, left-justified or right-justified. This looks mostly good (though you want to check your indentation a bit) but... > +static int __init mmp_sspa_modinit(void) > +{ > + audio_clk = clk_get(NULL, "mmp-audio"); > + if (IS_ERR(audio_clk)) > + return PTR_ERR(audio_clk); > + > + sysclk = clk_get(NULL, "mmp-sysclk"); > + if (IS_ERR(sysclk)) > + return PTR_ERR(sysclk); > + clk_enable(audio_clk); You should be doing this stuff from the driver, not from the module init (and ideally you'd be managing the audio clock dynamically). Even if it's the same clock every time in current silicon it's still a good idea to do this from a quality of implementation point of view.

2 1

Re: [alsa-devel] [PATCH v1 4/5] ASoC: add mmp brownstone support
by Mark Brown 08 Jun '12

08 Jun '12

On Mon, Jun 04, 2012 at 02:37:30PM +0800, Zhangfei Gao wrote: One small thing here... > + ret = snd_soc_add_card_controls(dapm->card, brownstone_dapm_control, > + ARRAY_SIZE(brownstone_dapm_control)); > + if (ret < 0) > + return ret; You can just set these up with the controls member of the card struct.

2 1

Re: [alsa-devel] [PATCH 0/2] HD-audio HDMI regression fixes with VGA-switcheroo
by Takashi Iwai 08 Jun '12

08 Jun '12

At Fri, 08 Jun 2012 13:26:57 +0200, Jörg-Volker Peetz wrote: > > Takashi Iwai wrote, on 06/07/12 12:15: > > Hi, > > > > this is a series of patches to fix the regressions of HD-audio HDMI > > on D-GPUs in 3.5-rc1 due to the support of VGA-switcheroo audio clients. > > > > The first patch adds a new helper function to vga-switcheroo and the > > second just uses that instead of an open code. > > > > Dave, if the first patch is OK, I'm going to apply it though sound tree. > > Let me know if any problem is found. > > > > Joerg, could you check whether this doesn't break your setup, too? > > > > > > thanks, > > > > Takashi > > Hello Takashi, > > I applied both your patches on 3.5-rc1. With this kernel I made three tests. > 1) After booting I switched off the discrete GPU via vga_switcheroo. > Then ca. 210 times the message > > hda-intel: spurious response 0x0:0x0, last cmd=0x170503 > > appears on the console and in the logs. > Starting X and using the built-in USB web-cam work, as well as sound with > mplayer2 works. > > 2) After booting I switched to the discrete GPU via > > echo -n DDIS > /sys/kernel/debugfs/vgaswitcheroo/switch > > Then I started X via startx. The desktop screen appears but the system > freezes. No reaction on keyboard or touchpad input. The computer is not > reachable via ethernet wire. Hm, these issues aren't seen on 3.5-rc1 before my patches? If so, we need to fix now. > > 3) After booting the machine freezes (hard-lock) when trying to switch to > dynamic power mangement on the integrated GPU via > > echo -n dynpm > /sys/class/drm/card0/device/power_method > > at least two trace calls are done. On the console only the last two lines of > one call trace and the last one are visible (machine is frozen). It says > (typed by hand, therefore, not complete): This is an irrelevant issue with my patches. thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC:isabelle:Fix warning message
by MR.Swami.Reddy＠ti.com 08 Jun '12

08 Jun '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> Patch to fix the below warning message: sound/soc/codecs/isabelle.c:1126: warning: 'isabelle_regmap' is used uninitialized in this function Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com> Signed-off-by: M R Swami Reddy <MR.Swami.Reddy(a)ti.com> --- sound/soc/codecs/isabelle.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index bcc77ef..a2efe1a 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1126,8 +1126,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, struct regmap *isabelle_regmap; int ret = 0; - i2c_set_clientdata(i2c, isabelle_regmap); - isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config); if (IS_ERR(isabelle_regmap)) { ret = PTR_ERR(isabelle_regmap);

3 2

[alsa-devel] [PATCH] ASoC: MAINTAINERS: Add maintainer for TI Isabelle Audio driver
by MR.Swami.Reddy＠ti.com 08 Jun '12

08 Jun '12

From: M R Swami Reddy <mr.swami.reddy(a)ti.com> Vishwas and I support the TI Isabelle audio driver. Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com> Signed-off-by: M R Swami Reddy <MR.Swami.Reddy(a)ti.com> --- MAINTAINERS | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 55f0fda..d052664 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6723,9 +6723,11 @@ F: include/linux/tifm.h TI LM49xxx FAMILY ASoC CODEC DRIVERS M: M R Swami Reddy <mr.swami.reddy(a)ti.com> +M: Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/lm49453* +F: sound/soc/codecs/isabelle* TI TWL4030 SERIES SOC CODEC DRIVER M: Peter Ujfalusi <peter.ujfalusi(a)ti.com>

3 2

[alsa-devel] [PATCH 0/2] HD-audio HDMI regression fixes with VGA-switcheroo
by Takashi Iwai 08 Jun '12

08 Jun '12

Hi, this is a series of patches to fix the regressions of HD-audio HDMI on D-GPUs in 3.5-rc1 due to the support of VGA-switcheroo audio clients. The first patch adds a new helper function to vga-switcheroo and the second just uses that instead of an open code. Dave, if the first patch is OK, I'm going to apply it though sound tree. Let me know if any problem is found. Joerg, could you check whether this doesn't break your setup, too? thanks, Takashi

2 4

[alsa-devel] ASoC updates for 3.5
by Mark Brown 08 Jun '12

08 Jun '12

The following changes since commit f8f5701bdaf9134b1f90e5044a82c66324d2073f: Linux 3.5-rc1 (2012-06-02 18:29:26 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.5 for you to fetch changes up to 3ff1ec27e9f567824e73316025a9025528494506: ASoC: wm2000: Always use a 4s timeout for the firmware (2012-06-08 07:00:13 +0800) ---------------------------------------------------------------- ASoC: Updates for 3.5 Some driver-specific updates, a few fixes for the dynamic PCM code and the addition of pinctrl support for the i.MX audmux which didn't make it into -rc1 due to cross tree dependency issues. ---------------------------------------------------------------- Liam Girdwood (3): ASoC: dapm: Fix connected widget capture path query. ASoC: dpcm: Fix dpcm_get_be() to check that DAI is BE ASoC: dapm: Fix input list to use source widgets Mark Brown (3): ASoC: wm8994: Ensure all AIFnCLK events are run from the _late variants ASoC: wm8994: Apply volume updates with clocks enabled ASoC: wm2000: Always use a 4s timeout for the firmware Richard Zhao (1): ASoC: imx-audmux: add pinctrl support sound/soc/codecs/wm2000.c | 59 +++++++++---------------- sound/soc/codecs/wm8994.c | 105 ++++++++++++++++++++++++++++---------------- sound/soc/fsl/imx-audmux.c | 8 ++++ sound/soc/soc-dapm.c | 4 +- sound/soc/soc-pcm.c | 6 +++ 5 files changed, 103 insertions(+), 79 deletions(-)

2 1

Re: [alsa-devel] 3.5-rc1 snd_usb_audio stream format breakage
by Takashi Iwai 08 Jun '12

08 Jun '12

At Thu, 7 Jun 2012 08:17:35 +0000, Jamie Heilman wrote: > > Takashi Iwai wrote: > > At Thu, 7 Jun 2012 03:38:30 +0000, > > Jamie Heilman wrote: > > > > > > With the USB audio streaming logic changes, my HeadRoom Bithead > > > (08bb:2902 Burr-Brown PCM2902) no longer works. Symptoms include the > > > loss of most of the normally available formats leaving only: > > > > > > root@cucamonga:~# cat /proc/asound/bithead/stream0 > > > Burr-Brown from TI USB Audio CODEC at usb-0000:00:1a.7-1.1.4, full speed : USB Audio > > > > > > Playback: > > > Status: Stop > > > Interface 1 > > > Altset 6 > > > Format: U8 > > > Channels: 1 > > > Endpoint: 2 OUT (ADAPTIVE) > > > Rates: 32000, 44100, 48000 > > > > > > Capture: > > > Status: Stop > > > Interface 2 > > > Altset 18 > > > Format: S8 > > > Channels: 1 > > > Endpoint: 4 IN (SYNC) > > > Rates: 11025 > > > > > > (normally 2 channel S16_LE is available, with capture rates up to > > > 48kHz ... only the last altset's are being reported now) and a slew of > > > proc registration warnings logged: > > > > > > ------------[ cut here ]------------ > > > WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6() > > > Hardware name: Precision WorkStation T3400 > > > proc_dir_entry 'card1/pcm0p' already registered > > > Modules linked in: snd_usb_audio(+) snd_usbmidi_lib snd_hda_intel(+) snd_hda_codec snd_hwdep snd_seq_midi snd_seq_midi_event snd_pcm snd_page_alloc snd_rawmidi snd_seq snd_seq_device snd_timer snd soundcore dcdbas floppy sr_mod usb_storage cdrom kvm_intel psmouse kvm tg3 evdev xfs dm_mod raid1 md_mod > > > Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2 > > > Call Trace: > > > [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c > > > [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48 > > > [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b > > > [<ffffffff811292f0>] proc_register+0x169/0x1a6 > > > [<ffffffff8112962e>] create_proc_entry+0x74/0x8c > > > [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd] > > > [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm] > > > [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio] > > > [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio] > > > [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb > > > [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio] > > > [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio] > > > [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio] > > > [<ffffffff812c49f5>] usb_probe_interface+0x136/0x1b1 > > > [<ffffffff81287e66>] ? driver_probe_device+0x1d3/0x1d3 > > > [<ffffffff81287d43>] driver_probe_device+0xb0/0x1d3 > > > [<ffffffff81287ec5>] __driver_attach+0x5f/0x83 > > > [<ffffffff812863a2>] bus_for_each_dev+0x54/0x86 > > > [<ffffffff81287730>] driver_attach+0x1e/0x20 > > > [<ffffffff8128747f>] bus_add_driver+0xfd/0x23a > > > [<ffffffff812882e8>] driver_register+0x93/0x10e > > > [<ffffffff812c383f>] usb_register_driver+0x7e/0x12a > > > [<ffffffffa0257000>] ? 0xffffffffa0256fff > > > [<ffffffffa0257000>] ? 0xffffffffa0256fff > > > [<ffffffffa0257040>] snd_usb_audio_init+0x40/0x1000 [snd_usb_audio] > > > [<ffffffff810001ff>] do_one_initcall+0x7f/0x134 > > > [<ffffffffa0257000>] ? 0xffffffffa0256fff > > > [<ffffffff810688d4>] sys_init_module+0x16cf/0x195e > > > [<ffffffff8106516c>] ? free_notes_attrs+0x4d/0x4d > > > [<ffffffff810c38cb>] ? mmap_region+0x1c7/0x469 > > > [<ffffffff813856d2>] system_call_fastpath+0x16/0x1b > > > ---[ end trace c1baea1a25a91564 ]--- > > > > Looks like the PCM substream assignment is buggy. > > Does the patch (untested at all) below help? > > Yep, that fixed it. Thanks! OK, applied the fix now. Thanks for quick testing. Takashi

1 0

Re: [alsa-devel] [PATCH v3 8/9] ASoC: tegra: add clk_prepare/clk_unprepare
by Mark Brown 08 Jun '12

08 Jun '12

On Tue, Jun 05, 2012 at 09:59:42AM +0530, Prashant Gaikwad wrote: > Use clk_prepare/clk_unprepare as required by the generic clk framework. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH] asoc: bfin: add i2s driver for blackfin bf60x processor
by Scott Jiang 08 Jun '12

08 Jun '12

This driver includes i2s dai, dma and sport driver. It enables i2s mode support on blackfin bf60x platform. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/Kconfig | 25 ++- sound/soc/blackfin/Makefile | 6 + sound/soc/blackfin/bf6xx-i2s-pcm.c | 294 +++++++++++++++++++++++++ sound/soc/blackfin/bf6xx-i2s.c | 245 +++++++++++++++++++++ sound/soc/blackfin/bf6xx-sport.c | 422 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf6xx-sport.h | 82 +++++++ 6 files changed, 1070 insertions(+), 4 deletions(-) create mode 100644 sound/soc/blackfin/bf6xx-i2s-pcm.c create mode 100644 sound/soc/blackfin/bf6xx-i2s.c create mode 100644 sound/soc/blackfin/bf6xx-sport.c create mode 100644 sound/soc/blackfin/bf6xx-sport.h diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 9f6bc55..2822b6a 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,6 +1,6 @@ config SND_BF5XX_I2S tristate "SoC I2S Audio for the ADI BF5xx chip" - depends on BLACKFIN + depends on BLACKFIN && !BF60x select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to @@ -8,10 +8,21 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. +config SND_BF6XX_I2S + tristate "SoC I2S Audio for the ADI BF6xx chip" + depends on BLACKFIN && BF60x + select SND_BF6XX_SOC_SPORT + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in I2S + mode (supports single stereo In/Out). + You will also need to select the audio interfaces to support below. + config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) - select SND_BF5XX_SOC_I2S + depends on (SND_BF5XX_I2S || SND_BF6XX_I2S) && (SPI_MASTER || I2C) + select SND_BF5XX_SOC_I2S if SND_BF5XX_I2S + select SND_BF6XX_SOC_I2S if SND_BF6XX_I2S select SND_SOC_SSM2602 help Say Y if you want to add support for SoC audio on BF527-EZKIT. @@ -162,9 +173,15 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SOC_SPORT tristate +config SND_BF6XX_SOC_SPORT + tristate + config SND_BF5XX_SOC_I2S tristate +config SND_BF6XX_SOC_I2S + tristate + config SND_BF5XX_SOC_TDM tristate @@ -173,7 +190,7 @@ config SND_BF5XX_SOC_AC97 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) + depends on (BLACKFIN && SND_SOC) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 1bf86cc..a05335a 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,18 +1,24 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o +snd-bf6xx-i2s-objs := bf6xx-i2s-pcm.o snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o +snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o +obj-$(CONFIG_SND_BF6XX_I2S) += snd-bf6xx-i2s.o obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o +obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support diff --git a/sound/soc/blackfin/bf6xx-i2s-pcm.c b/sound/soc/blackfin/bf6xx-i2s-pcm.c new file mode 100644 index 0000000..cc3216f --- /dev/null +++ b/sound/soc/blackfin/bf6xx-i2s-pcm.c @@ -0,0 +1,294 @@ +/* + * bf6xx-i2s-pcm.c - Analog Devices BF6XX i2s dma driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/mm.h> +#include <linux/types.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "bf6xx-sport.h" + + +#include <linux/dma-mapping.h> +#include <linux/gfp.h> +#include <asm/dma.h> + +static void bfin_dma_irq(void *data) +{ + struct snd_pcm_substream *pcm = data; + snd_pcm_period_elapsed(pcm); +} + +static const struct snd_pcm_hardware bfin_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 32, + .period_bytes_max = 0x10000, + .periods_min = 1, + .periods_max = PAGE_SIZE/32, + .buffer_bytes_max = 0x20000, /* 128 kbytes */ + .fifo_size = 16, +}; + +static int bfin_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + size_t size = bfin_pcm_hardware.buffer_bytes_max; + snd_pcm_lib_malloc_pages(substream, size); + + return 0; +} + +static int bfin_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int bfin_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int period_bytes = frames_to_bytes(runtime, runtime->period_size); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport_set_tx_callback(sport, bfin_dma_irq, substream); + sport_config_tx_dma(sport, runtime->dma_area, + runtime->periods, period_bytes); + } else { + sport_set_rx_callback(sport, bfin_dma_irq, substream); + sport_config_rx_dma(sport, runtime->dma_area, + runtime->periods, period_bytes); + } + + return 0; +} + +static int bfin_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_start(sport); + else + sport_rx_start(sport); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_stop(sport); + else + sport_rx_stop(sport); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t bfin_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + unsigned int diff; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + diff = sport_curr_offset_tx(sport); + } else { + diff = sport_curr_offset_rx(sport); + } + + /* + * TX at least can report one frame beyond the end of the + * buffer if we hit the wraparound case - clamp to within the + * buffer as the ALSA APIs require. + */ + if (diff == snd_pcm_lib_buffer_bytes(substream)) + diff = 0; + + frames = bytes_to_frames(substream->runtime, diff); + + return frames; +} + +static int bfin_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + snd_soc_set_runtime_hwparams(substream, &bfin_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + runtime->private_data = sport; + return 0; +} + +static int bfin_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + size_t size = vma->vm_end - vma->vm_start; + vma->vm_start = (unsigned long)runtime->dma_area; + vma->vm_end = vma->vm_start + size; + vma->vm_flags |= VM_SHARED; + + return 0 ; +} + +static struct snd_pcm_ops bfin_pcm_i2s_ops = { + .open = bfin_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bfin_pcm_hw_params, + .hw_free = bfin_pcm_hw_free, + .prepare = bfin_pcm_prepare, + .trigger = bfin_pcm_trigger, + .pointer = bfin_pcm_pointer, + .mmap = bfin_pcm_mmap, +}; + +static int bfin_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bfin_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static void bfin_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(NULL, buf->bytes, buf->area, 0); + buf->area = NULL; + } +} + +static u64 bfin_pcm_dmamask = DMA_BIT_MASK(32); + +int bfin_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &bfin_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) { + ret = bfin_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->driver->capture.channels_min) + ret = bfin_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + return ret; +} + +static struct snd_soc_platform_driver bfin_i2s_soc_platform = { + .ops = &bfin_pcm_i2s_ops, + .pcm_new = bfin_pcm_i2s_new, + .pcm_free = bfin_pcm_free_dma_buffers, +}; + +static int __devinit bfin_i2s_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &bfin_i2s_soc_platform); +} + +static int __devexit bfin_i2s_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver bfin_i2s_pcm_driver = { + .driver = { + .name = "bfin-i2s-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = bfin_i2s_soc_platform_probe, + .remove = __devexit_p(bfin_i2s_soc_platform_remove), +}; + +static int __init snd_bfin_i2s_pcm_init(void) +{ + return platform_driver_register(&bfin_i2s_pcm_driver); +} +module_init(snd_bfin_i2s_pcm_init); + +static void __exit snd_bfin_i2s_pcm_exit(void) +{ + platform_driver_unregister(&bfin_i2s_pcm_driver); +} +module_exit(snd_bfin_i2s_pcm_exit); + +MODULE_DESCRIPTION("Analog Devices BF6XX i2s dma driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c new file mode 100644 index 0000000..c4a5499 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -0,0 +1,245 @@ +/* + * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "bf6xx-sport.h" + +struct sport_params param; + +static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR + | SPORT_CTL_LFS | SPORT_CTL_LAFS); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE + | SPORT_CTL_LFS; + break; + case SND_SOC_DAIFMT_DSP_A: + param.spctl |= SPORT_CTL_FSR; + break; + case SND_SOC_DAIFMT_LEFT_J: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS + | SPORT_CTL_LAFS; + break; + default: + dev_err(dev, "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_err(dev, "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~SPORT_CTL_SLEN; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + param.spctl |= 0x70; + sport->wdsize = 1; + case SNDRV_PCM_FORMAT_S16_LE: + param.spctl |= 0xf0; + sport->wdsize = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + param.spctl |= 0x170; + sport->wdsize = 3; + break; + case SNDRV_PCM_FORMAT_S32_LE: + param.spctl |= 0x1f0; + sport->wdsize = 4; + break; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = sport_set_tx_params(sport, &param); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + } else { + ret = sport_set_rx_params(sport, &param); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + } + return 0; +} + +#ifdef CONFIG_PM +static int bfin_i2s_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_active) + sport_rx_stop(sport); + if (dai->playback_active) + sport_tx_stop(sport); + return 0; +} + +static int bfin_i2s_resume(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret; + + ret = sport_set_tx_params(sport, &param); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + ret = sport_set_rx_params(sport, &param); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + + return 0; +} + +#else +#define bfin_i2s_suspend NULL +#define bfin_i2s_resume NULL +#endif + +#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops bfin_i2s_dai_ops = { + .hw_params = bfin_i2s_hw_params, + .set_fmt = bfin_i2s_set_dai_fmt, +}; + +static struct snd_soc_dai_driver bfin_i2s_dai = { + .suspend = bfin_i2s_suspend, + .resume = bfin_i2s_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .ops = &bfin_i2s_dai_ops, +}; + +static int __devinit bfin_i2s_probe(struct platform_device *pdev) +{ + struct sport_device *sport; + struct device *dev = &pdev->dev; + int ret; + + sport = sport_create(pdev); + if (!sport) + return -ENODEV; + + /* register with the ASoC layers */ + ret = snd_soc_register_dai(dev, &bfin_i2s_dai); + if (ret) { + dev_err(dev, "Failed to register DAI: %d\n", ret); + sport_delete(sport); + return ret; + } + platform_set_drvdata(pdev, sport); + + return 0; +} + +static int __devexit bfin_i2s_remove(struct platform_device *pdev) +{ + struct sport_device *sport = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + sport_delete(sport); + + return 0; +} + +static struct platform_driver bfin_i2s_driver = { + .probe = bfin_i2s_probe, + .remove = __devexit_p(bfin_i2s_remove), + .driver = { + .name = "bfin-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init bfin_i2s_init(void) +{ + return platform_driver_register(&bfin_i2s_driver); +} + +static void __exit bfin_i2s_exit(void) +{ + platform_driver_unregister(&bfin_i2s_driver); +} + +module_init(bfin_i2s_init); +module_exit(bfin_i2s_exit); + +MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c new file mode 100644 index 0000000..ea4b844 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -0,0 +1,422 @@ +/* + * bf6xx_sport.c Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <asm/blackfin.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "bf6xx-sport.h" + +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; + sport->tx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_tx_params); + +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; + sport->rx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_rx_params); + +static int compute_wdsize(size_t wdsize) +{ + switch (wdsize) { + case 1: + return WDSIZE_8 | PSIZE_8; + case 2: + return WDSIZE_16 | PSIZE_16; + default: + return WDSIZE_32 | PSIZE_32; + } +} + +void sport_tx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); + set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->tx_dma_chan); + sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_tx_start); + +void sport_rx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); + set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->rx_dma_chan); + sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_rx_start); + +void sport_tx_stop(struct sport_device *sport) +{ + sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->tx_dma_chan); +} +EXPORT_SYMBOL(sport_tx_stop); + +void sport_rx_stop(struct sport_device *sport) +{ + sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->rx_dma_chan); +} +EXPORT_SYMBOL(sport_rx_stop); + +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data) +{ + sport->tx_callback = tx_callback; + sport->tx_data = tx_data; +} +EXPORT_SYMBOL(sport_set_tx_callback); + +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data) +{ + sport->rx_callback = rx_callback; + sport->rx_data = rx_data; +} +EXPORT_SYMBOL(sport_set_rx_callback); + +static void setup_desc(struct dmasg *desc, void *buf, int fragcount, + size_t fragsize, unsigned int cfg, + unsigned int count, size_t wdsize) +{ + + int i; + + for (i = 0; i < fragcount; ++i) { + desc[i].next_desc_addr = &(desc[i + 1]); + desc[i].start_addr = (unsigned long)buf + i*fragsize; + desc[i].cfg = cfg; + desc[i].x_count = count; + desc[i].x_modify = wdsize; + desc[i].y_count = 0; + desc[i].y_modify = 0; + } + + /* make circular */ + desc[fragcount-1].next_desc_addr = desc; +} + +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + + sport->tx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->tx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->tx_desc) + return -ENOMEM; + + sport->tx_buf = buf; + sport->tx_fragsize = fragsize; + sport->tx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; + + setup_desc(sport->tx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_tx_dma); + +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); + + sport->rx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->rx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->rx_desc) + return -ENOMEM; + + sport->rx_buf = buf; + sport->rx_fragsize = fragsize; + sport->rx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) + | WNR | NDSIZE_6; + + setup_desc(sport->rx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_rx_dma); + +unsigned long sport_curr_offset_tx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); + + return (unsigned char *)curr - sport->tx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_tx); + +unsigned long sport_curr_offset_rx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); + + return (unsigned char *)curr - sport->rx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_rx); + +static irqreturn_t sport_tx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + static unsigned long status; + + status = get_dma_curr_irqstat(sport->tx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->tx_dma_chan); + SSYNC(); + } + if (sport->tx_callback) + sport->tx_callback(sport->tx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_rx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + unsigned long status; + + status = get_dma_curr_irqstat(sport->rx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->rx_dma_chan); + SSYNC(); + } + if (sport->rx_callback) + sport->rx_callback(sport->rx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_err_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + struct device *dev = &sport->pdev->dev; + + if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: TUVF\n"); + if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: ROVF\n"); + + return IRQ_HANDLED; +} + +static int sport_get_resource(struct sport_device *sport) +{ + struct platform_device *pdev = sport->pdev; + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + if (!pdata) { + dev_err(dev, "No platform data\n"); + return -ENODEV; + } + sport->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "No tx MEM resource\n"); + return -ENODEV; + } + sport->tx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + dev_err(dev, "No rx MEM resource\n"); + return -ENODEV; + } + sport->rx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "No tx DMA resource\n"); + return -ENODEV; + } + sport->tx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "No rx DMA resource\n"); + return -ENODEV; + } + sport->rx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "No tx error irq resource\n"); + return -ENODEV; + } + sport->tx_err_irq = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); + if (!res) { + dev_err(dev, "No rx error irq resource\n"); + return -ENODEV; + } + sport->rx_err_irq = res->start; + + return 0; +} + +static int sport_request_resource(struct sport_device *sport) +{ + struct device *dev = &sport->pdev->dev; + int ret; + + ret = peripheral_request_list(sport->pin_req, "soc-audio"); + if (ret) { + dev_err(dev, "Unable to request sport pin\n"); + return ret; + } + + ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); + goto err_tx_dma; + } + set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); + + ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); + goto err_rx_dma; + } + set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); + + ret = request_irq(sport->tx_err_irq, sport_err_irq, + 0, "SPORT TX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); + goto err_tx_irq; + } + + ret = request_irq(sport->rx_err_irq, sport_err_irq, + 0, "SPORT RX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); + goto err_rx_irq; + } + + return 0; +err_rx_irq: + free_irq(sport->tx_err_irq, sport); +err_tx_irq: + free_dma(sport->rx_dma_chan); +err_rx_dma: + free_dma(sport->tx_dma_chan); +err_tx_dma: + peripheral_free_list(sport->pin_req); + return ret; +} + +static void sport_free_resource(struct sport_device *sport) +{ + free_irq(sport->rx_err_irq, sport); + free_irq(sport->tx_err_irq, sport); + free_dma(sport->rx_dma_chan); + free_dma(sport->tx_dma_chan); + peripheral_free_list(sport->pin_req); +} + +struct sport_device *sport_create(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct sport_device *sport; + int ret; + + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "Unable to allocate memory for sport device\n"); + return NULL; + } + sport->pdev = pdev; + + ret = sport_get_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + ret = sport_request_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + dev_info(dev, "SPORT create success\n"); + return sport; +} +EXPORT_SYMBOL(sport_create); + +void sport_delete(struct sport_device *sport) +{ + sport_free_resource(sport); +} +EXPORT_SYMBOL(sport_delete); + +MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h new file mode 100644 index 0000000..307d193 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.h @@ -0,0 +1,82 @@ +/* + * bf6xx_sport - Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef _BF6XX_SPORT_H_ +#define _BF6XX_SPORT_H_ + +#include <linux/platform_device.h> +#include <asm/bfin_sport3.h> + +struct sport_device { + struct platform_device *pdev; + const unsigned short *pin_req; + struct sport_register *tx_regs; + struct sport_register *rx_regs; + int tx_dma_chan; + int rx_dma_chan; + int tx_err_irq; + int rx_err_irq; + + void (*tx_callback)(void *data); + void *tx_data; + void (*rx_callback)(void *data); + void *rx_data; + + struct dmasg *tx_desc; + struct dmasg *rx_desc; + unsigned int tx_desc_size; + unsigned int rx_desc_size; + unsigned char *tx_buf; + unsigned char *rx_buf; + unsigned int tx_fragsize; + unsigned int rx_fragsize; + unsigned int tx_frags; + unsigned int rx_frags; + unsigned int wdsize; +}; + +struct sport_params { + u32 spctl; + u32 div; +}; + +struct sport_device *sport_create(struct platform_device *pdev); +void sport_delete(struct sport_device *sport); +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params); +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params); +void sport_tx_start(struct sport_device *sport); +void sport_rx_start(struct sport_device *sport); +void sport_tx_stop(struct sport_device *sport); +void sport_rx_stop(struct sport_device *sport); +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data); +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data); +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +unsigned long sport_curr_offset_tx(struct sport_device *sport); +unsigned long sport_curr_offset_rx(struct sport_device *sport); + + + +#endif -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ASoC: tegra: add MODULE_DEVICE_TABLE to tegra30_ahub
by Stephen Warren 08 Jun '12

08 Jun '12

From: Stephen Warren <swarren(a)nvidia.com> This allows the module to automatically load when instantiated from device tree. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This is useful in 3.5. sound/soc/tegra/tegra30_ahub.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 57cd419..f43edb3 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -629,3 +629,4 @@ MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); MODULE_DESCRIPTION("Tegra30 AHUB driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra30_ahub_of_match); -- 1.7.0.4

2 1

Re: [alsa-devel] [PATCH 3/3] ASoC: follow the updated samsung DMA common operations
by Mark Brown 08 Jun '12

08 Jun '12

On Tue, Jun 05, 2012 at 04:20:10PM +0900, Boojin Kim wrote: > This patch uses config() function to configure DMA transmit options. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> It'd be good if we could get all the older Samsung platforms moved over to dmaengine...

1 0

[alsa-devel] [patch] ASoC: isabelle: using an uninitialized variable
by Dan Carpenter 08 Jun '12

08 Jun '12

We should set "isabelle_regmap" before using it. GCC complains. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index bcc77ef..0d62f3b 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1126,8 +1126,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, struct regmap *isabelle_regmap; int ret = 0; - i2c_set_clientdata(i2c, isabelle_regmap); - isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config); if (IS_ERR(isabelle_regmap)) { ret = PTR_ERR(isabelle_regmap); @@ -1135,6 +1133,7 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, ret); return ret; } + i2c_set_clientdata(i2c, isabelle_regmap); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_isabelle, isabelle_dai,

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix input list to use source widgets.
by Liam Girdwood 08 Jun '12

08 Jun '12

We should only add source widgets to the input list. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9a67a75..20b3402 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -928,7 +928,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, /* do we need to add this widget to the list ? */ if (list) { int err; - err = dapm_list_add_widget(list, path->sink); + err = dapm_list_add_widget(list, path->source); if (err < 0) { dev_err(widget->dapm->dev, "could not add widget %s\n", widget->name); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Fix dpcm_get_be() to check that DAI is BE.
by Liam Girdwood 08 Jun '12

08 Jun '12

Make sure that the dpcm_get_be() only returns BE DAI links. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index edb0d0c..1b6f6be 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -830,6 +830,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, for (i = 0; i < card->num_links; i++) { be = &card->rtd[i]; + if (!be->dai_link->no_pcm) + continue; + if (be->cpu_dai->playback_widget == widget || be->codec_dai->playback_widget == widget) return be; @@ -839,6 +842,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, for (i = 0; i < card->num_links; i++) { be = &card->rtd[i]; + if (!be->dai_link->no_pcm) + continue; + if (be->cpu_dai->capture_widget == widget || be->codec_dai->capture_widget == widget) return be; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm2000: Always use a 4s timeout for the firmware
by Mark Brown 08 Jun '12

08 Jun '12

Rather than having varying timeouts depending on the transition always use a 4s timeout. This provides better diagnostics for clocking errors and ensures compatibility with current calibration firmwares. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 59 ++++++++++++++++----------------------------- 1 file changed, 21 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 78a148f..3fd5b29 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -99,8 +99,9 @@ static void wm2000_reset(struct wm2000_priv *wm2000) } static int wm2000_poll_bit(struct i2c_client *i2c, - unsigned int reg, u8 mask, int timeout) + unsigned int reg, u8 mask) { + int timeout = 4000; int val; val = wm2000_read(i2c, reg); @@ -119,7 +120,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c, static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int ret, timeout; + int ret; BUG_ON(wm2000->anc_mode != ANC_OFF); @@ -140,13 +141,13 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) /* Wait for ANC engine to become ready */ if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, - WM2000_ANC_ENG_IDLE, 1)) { + WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, "ANC engine failed to reset\n"); return -ETIMEDOUT; } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_BOOT_COMPLETE, 1)) { + WM2000_STATUS_BOOT_COMPLETE)) { dev_err(&i2c->dev, "ANC engine failed to initialise\n"); return -ETIMEDOUT; } @@ -173,16 +174,13 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) dev_dbg(&i2c->dev, "Download complete\n"); if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_MOUSE_ENABLE | WM2000_MODE_THERMAL_ENABLE); } else { - timeout = 10; - wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_MOUSE_ENABLE | WM2000_MODE_THERMAL_ENABLE); @@ -201,9 +199,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_MOUSE_ACTIVE, timeout)) { - dev_err(&i2c->dev, "Timed out waiting for device after %dms\n", - timeout * 10); + WM2000_STATUS_MOUSE_ACTIVE)) { + dev_err(&i2c->dev, "Timed out waiting for device\n"); return -ETIMEDOUT; } @@ -218,28 +215,25 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) static int wm2000_power_down(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int timeout; if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_POWER_DOWN); } else { - timeout = 10; wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_POWER_DOWN); } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_POWER_DOWN_COMPLETE, timeout)) { + WM2000_STATUS_POWER_DOWN_COMPLETE)) { dev_err(&i2c->dev, "Timeout waiting for ANC power down\n"); return -ETIMEDOUT; } if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, - WM2000_ANC_ENG_IDLE, 1)) { + WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); return -ETIMEDOUT; } @@ -268,13 +262,13 @@ static int wm2000_enter_bypass(struct i2c_client *i2c, int analogue) } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_ANC_DISABLED, 10)) { + WM2000_STATUS_ANC_DISABLED)) { dev_err(&i2c->dev, "Timeout waiting for ANC disable\n"); return -ETIMEDOUT; } if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, - WM2000_ANC_ENG_IDLE, 1)) { + WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); return -ETIMEDOUT; } @@ -311,7 +305,7 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_MOUSE_ACTIVE, 10)) { + WM2000_STATUS_MOUSE_ACTIVE)) { dev_err(&i2c->dev, "Timed out waiting for MOUSE\n"); return -ETIMEDOUT; } @@ -325,38 +319,32 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int timeout; BUG_ON(wm2000->anc_mode != ANC_ACTIVE); if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_STANDBY_ENTRY); } else { - timeout = 10; - wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_STANDBY_ENTRY); } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_ANC_DISABLED, timeout)) { + WM2000_STATUS_ANC_DISABLED)) { dev_err(&i2c->dev, "Timed out waiting for ANC disable after 1ms\n"); return -ETIMEDOUT; } - if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE, - 1)) { + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, - "Timed out waiting for standby after %dms\n", - timeout * 10); + "Timed out waiting for standby\n"); return -ETIMEDOUT; } @@ -374,23 +362,19 @@ static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int timeout; BUG_ON(wm2000->anc_mode != ANC_STANDBY); wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_MOUSE_ENABLE); } else { - timeout = 10; - wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_MOUSE_ENABLE); @@ -400,9 +384,8 @@ static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_MOUSE_ACTIVE, timeout)) { - dev_err(&i2c->dev, "Timed out waiting for MOUSE after %dms\n", - timeout * 10); + WM2000_STATUS_MOUSE_ACTIVE)) { + dev_err(&i2c->dev, "Timed out waiting for MOUSE\n"); return -ETIMEDOUT; } -- 1.7.10

1 0

[alsa-devel] [Q] Writing an alsa module for usb audio & video capture device
by Ezequiel Garcia 07 Jun '12

07 Jun '12

Hi everyone, I'm writing a driver for stk1160 usb bridge (audio & video capture device). The video part is almost done [1] and now I've started the alsa part. As the driver replaces an in-tree staging driver I could just grab the alsa part from the original driver (called easycap), or look into em28xx sources. But as I want to learn a thing or two in this life, I've decided to take a look at sound drivers first. Currently I'm reading through sound/usb/6fire driver (picked randomly). I would like to know: do you have any suggestions and/or hints in order to write a *nice* driver? What driver would you consider as being a *correct* usb sound capture implementation? [1] http://patchwork.linuxtv.org/patch/11577/ Thanks, Ezequiel.

2 4

[alsa-devel] Internal Mics and Speakers not visible in PulseAudio
by David Henningsson 07 Jun '12

07 Jun '12

Posted to both alsa-devel and pulseaudio-discuss lists, as this is a cross issue. PulseAudio tries to figure out what equipment is present on the machine using the mixer. If it finds e g "Headphone Playback Volume", "Headphone Playback Switch", or "Headphone Jack", it assumes that a headphone is present and creates a headphone port. If it finds no ports, it creates a fallback "Analog Output" port. Now assume that we have a laptop with speakers and a headphone jack, but with only a "Master Playback Volume", and a "Headphone Jack" for the jack detection. PulseAudio will take the "Headphone Jack", and create a headphone port. Now, if this port is currently unconnected/unavailable, it will not show up at all [1]. As a result, the internal speakers - for which no port was created - will be essentially unusable. Now, I thought this was a problem with a pair of machines only, that we could easily quirk on the PulseAudio side. But after looking through Ubuntu bugs and posting a blog post [2] about the issue, I have now collected 21 different machines affected, mostly on the input side. Several Realtek chips are affected on the dmic side, and some older STAC92xx chips have problems in both directions. So it becomes evident to me, that this is something that needs to be fixed pretty fast. So, to move forward, we need to expose these speakers and internal mics to userspace. A very simple (untested) patch is attached. This would make "Internal Mic Jack" and "Speaker Jack" show up. Note that the actual status can be overridden/ignored on the PulseAudio side - the importance here is the sign that there are internal mics and speakers, so that the PulseAudio ports get created. I could develop it a little to make sure we don't actually do any jack detection for these pins but always return them as being true/present. That is, if you approve the idea? I admit that the "Internal Mic Jack" is somewhat misleading/hacky as the internal mic is not connected to any physical jack, but it seems like the simplest way of resolving the problem currently. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] At least in Ubuntu's new Sound Settings UI. [2] http://voices.canonical.com/david.henningsson/2012/05/22/three-audio-bugs-i…

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[alsa-devel] [PATCH V2] ALSA: hda - power setting error check
by Wang Xingchao 07 Jun '12

07 Jun '12

codec may reject power state transition requests(reporting PS-ERROR set), in that case we re-issue a power state setting and check error bit again. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d0ca370..a93c7ca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3526,6 +3526,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { + int state, count = 0; if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3537,9 +3538,19 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); msleep(epss? 10:100); } - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + + /* repeat power states setting at most 10 times*/ + while (count++ > 10) { + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (!(state & AC_PWRST_ERROR)) + break; + } + if (count > 10) + snd_printk(KERN_WARNING "Power states transition reject!\n"); } /* modem wake on ring: transition from D2 to D0 in less than 2ms. For modems, -- 1.7.9.5

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Re: [alsa-devel] [PATCH v1 1/5] dmaengine: mmp_tdma: add mmp tdma support
by Vinod Koul 07 Jun '12

07 Jun '12

On Mon, 2012-06-04 at 14:37 +0800, Zhangfei Gao wrote: > Add support for two-channel dma under dmaengine > support: mmp-adma and pxa910-squ > > Signed-off-by: Zhangfei Gao <zhangfei.gao(a)marvell.com> > Signed-off-by: Leo Yan <leoy(a)marvell.com> > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> > + > +static int mmp_tdma_clear_chan_irq(struct mmp_tdma_chan *tdmac) > +{ > + u32 reg = readl(tdmac->reg_base + TDISR); > + > + if (reg & TDISR_COMP) { > + /* clear irq */ > + reg &= ~TDISR_COMP; > + writel(reg, tdmac->reg_base + TDISR); > + > + return 1; hrmm.? > + } > + return 0; > +} > + > +static irqreturn_t mmp_tdma_chan_handler(int irq, void *dev_id) > +{ > + struct mmp_tdma_chan *tdmac = dev_id; > + > + if (mmp_tdma_clear_chan_irq(tdmac)) { > + tdmac->pos = (tdmac->pos + tdmac->period_len) % tdmac->buf_len; > + tasklet_schedule(&tdmac->tasklet); > + return IRQ_HANDLED; > + } else > + return IRQ_NONE; > +} > + > +static irqreturn_t mmp_tdma_int_handler(int irq, void *dev_id) > +{ > + struct mmp_tdma_device *tdev = dev_id; > + int i, ret; > + int irq_num = 0; > + > + for (i = 0; i < TDMA_CHANNEL_NUM; i++) { > + struct mmp_tdma_chan *tdmac = tdev->tdmac[i]; > + > + ret = mmp_tdma_chan_handler(irq, tdmac); > + if (ret == IRQ_HANDLED) > + irq_num++; > + } > + > + if (irq_num) > + return IRQ_HANDLED; > + else > + return IRQ_NONE; > +} > + > +static void dma_do_tasklet(unsigned long data) > +{ > + struct mmp_tdma_chan *tdmac = (struct mmp_tdma_chan *)data; > + > + if (tdmac->desc.callback) > + tdmac->desc.callback(tdmac->desc.callback_param); > + > +} > + > +static void mmp_tdma_free_descriptor(struct mmp_tdma_chan *tdmac) > +{ > + struct gen_pool *gpool; > + int size = tdmac->desc_num * sizeof(struct mmp_tdma_desc); > + > + gpool = sram_get_gpool("asram"); > + if (tdmac->desc_arr) > + gen_pool_free(gpool, (unsigned long)tdmac->desc_arr, > + size); > + tdmac->desc_arr = NULL; > + > + return; > +} > + > +static dma_cookie_t mmp_tdma_tx_submit(struct dma_async_tx_descriptor *tx) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(tx->chan); > + > + mmp_tdma_chan_set_desc(tdmac, tdmac->desc_arr_phys); This seems odd. In submit you are supposed to move this to your pending list. Btw where is the .issue_pending handler? > + > + return 0; > +} > + > +static int mmp_tdma_alloc_chan_resources(struct dma_chan *chan) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(chan); > + int ret; > + > + dev_dbg(tdmac->dev, "%s: enter\n", __func__); > + > + dma_async_tx_descriptor_init(&tdmac->desc, chan); > + tdmac->desc.tx_submit = mmp_tdma_tx_submit; > + > + if (tdmac->irq) { > + ret = devm_request_irq(tdmac->dev, tdmac->irq, > + mmp_tdma_chan_handler, IRQF_DISABLED, "tdma", tdmac); > + if (ret) > + goto err_request_irq; > + } > + > + return 0; NO. This is supposed to return the number of descriptors allocated. > + > +err_request_irq: > + mmp_tdma_free_descriptor(tdmac); > + return ret; > +} > + > +static void mmp_tdma_free_chan_resources(struct dma_chan *chan) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(chan); > + > + dev_dbg(tdmac->dev, "%s: enter\n", __func__); > + > + if (tdmac->irq) > + devm_free_irq(tdmac->dev, tdmac->irq, tdmac); > + mmp_tdma_disable_chan(tdmac); channel should be already disabled, why do you need this here? > + mmp_tdma_free_descriptor(tdmac); > + return; > +} > + > + > +static struct dma_async_tx_descriptor *mmp_tdma_prep_slave_sg( > + struct dma_chan *chan, struct scatterlist *sgl, > + unsigned int sg_len, enum dma_transfer_direction direction, > + unsigned long flags, void *context) > +{ > + /* > + * This operation is not supported on the TDMA controller > + * > + * However, we need to provide the function pointer to allow the > + * device_control() method to work. > + */ > + return NULL; > +} Pls remove if not supported > + > +static int mmp_tdma_alloc_descriptor(struct mmp_tdma_chan *tdmac) > +{ > + struct gen_pool *gpool; > + int size = tdmac->desc_num * sizeof(struct mmp_tdma_desc); > + > + dev_dbg(tdmac->dev, "%s: enter\n", __func__); > + > + gpool = sram_get_gpool("asram"); > + if (!gpool) > + return -ENOMEM; > + > + tdmac->desc_arr = (void *)gen_pool_alloc(gpool, size); > + if (!tdmac->desc_arr) > + return -ENOMEM; > + > + tdmac->desc_arr_phys = gen_pool_virt_to_phys(gpool, > + (unsigned long)tdmac->desc_arr); > + > + return 0; this needs fix > +} > + > +static struct dma_async_tx_descriptor *mmp_tdma_prep_dma_cyclic( > + struct dma_chan *chan, dma_addr_t dma_addr, size_t buf_len, > + size_t period_len, enum dma_transfer_direction direction, > + void *context) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(chan); > + int num_periods = buf_len / period_len; > + int i = 0, buf = 0; > + int ret; > + > + if (tdmac->status != DMA_SUCCESS) > + return NULL; > + > + if (period_len > TDMA_MAX_XFER_BYTES) { > + dev_err(tdmac->dev, > + "maximum period size exceeded: %d > %d\n", > + period_len, TDMA_MAX_XFER_BYTES); > + goto err_out; > + } > + > + tdmac->status = DMA_IN_PROGRESS; ??? > + tdmac->desc_num = num_periods; > + ret = mmp_tdma_alloc_descriptor(tdmac); > + if (ret < 0) > + goto err_out; > + > + while (buf < buf_len) { > + struct mmp_tdma_desc *desc = &tdmac->desc_arr[i]; what if i call prepare twice on the same channel? Better way would to manage a descriptor list, see other drivers for examples. > + > + if (i + 1 == num_periods) > + desc->nxt_desc = tdmac->desc_arr_phys; > + else > + desc->nxt_desc = tdmac->desc_arr_phys + > + sizeof(*desc) * (i + 1); pls use kernel link list, it is provided to you so that you dont have to do above. > + > + if (direction == DMA_MEM_TO_DEV) { > + desc->src_addr = dma_addr; > + desc->dst_addr = tdmac->dev_addr; > + } else { > + desc->src_addr = tdmac->dev_addr; > + desc->dst_addr = dma_addr; > + } > + desc->byte_cnt = period_len; > + dma_addr += period_len; > + buf += period_len; > + i++; > + } > + > + tdmac->buf_len = buf_len; > + tdmac->period_len = period_len; > + tdmac->pos = 0; > + > + return &tdmac->desc; > + > +err_out: > + tdmac->status = DMA_ERROR; > + return NULL; > +} > + > +static int mmp_tdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, > + unsigned long arg) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(chan); > + struct dma_slave_config *dmaengine_cfg = (void *)arg; > + int ret = 0; > + > + switch (cmd) { > + case DMA_TERMINATE_ALL: > + mmp_tdma_disable_chan(tdmac); > + break; > + case DMA_PAUSE: > + mmp_tdma_pause_chan(tdmac); > + break; > + case DMA_RESUME: > + mmp_tdma_resume_chan(tdmac); > + break; > + case DMA_SLAVE_CONFIG: > + if (dmaengine_cfg->direction == DMA_DEV_TO_MEM) { > + tdmac->dev_addr = dmaengine_cfg->src_addr; > + tdmac->burst_sz = dmaengine_cfg->src_maxburst; > + } else { > + tdmac->dev_addr = dmaengine_cfg->dst_addr; > + tdmac->burst_sz = dmaengine_cfg->dst_maxburst; > + } > + tdmac->dir = dmaengine_cfg->direction; > + return mmp_tdma_config_chan(tdmac); > + default: > + ret = -ENOSYS; > + } > + > + return ret; > +} > + > +static enum dma_status mmp_tdma_tx_status(struct dma_chan *chan, > + dma_cookie_t cookie, struct dma_tx_state *txstate) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(chan); > + > + dma_set_residue(txstate, tdmac->buf_len - tdmac->pos); > + > + return tdmac->status; > +} > + > +static void mmp_tdma_issue_pending(struct dma_chan *chan) > +{ > + struct mmp_tdma_chan *tdmac = to_mmp_tdma_chan(chan); > + > + mmp_tdma_enable_chan(tdmac); can you queue the descriptors here? That was purpose to have separate issue pending and submit. > +} > + > +static int __devexit mmp_tdma_remove(struct platform_device *pdev) > +{ > + struct mmp_tdma_device *tdev = platform_get_drvdata(pdev); > + > + dma_async_device_unregister(&tdev->device); > + return 0; > +} > + > +static int __devinit mmp_tdma_chan_init(struct mmp_tdma_device *tdev, > + int idx, int irq, int type) > +{ > + struct mmp_tdma_chan *tdmac; > + > + if (idx >= TDMA_CHANNEL_NUM) { > + dev_err(tdev->dev, "too many channels for device!\n"); > + return -EINVAL; > + } > + > + /* alloc channel */ > + tdmac = devm_kzalloc(tdev->dev, sizeof(*tdmac), GFP_KERNEL); > + if (!tdmac) { > + dev_err(tdev->dev, "no free memory for DMA channels!\n"); > + return -ENOMEM; > + } > + if (irq) > + tdmac->irq = irq + idx; > + tdmac->dev = tdev->dev; > + tdmac->chan.device = &tdev->device; > + tdmac->idx = idx; > + tdmac->type = type; > + tdmac->reg_base = (unsigned long)tdev->base + idx * 4; > + tdmac->status = DMA_SUCCESS; > + tdev->tdmac[tdmac->idx] = tdmac; > + tasklet_init(&tdmac->tasklet, dma_do_tasklet, (unsigned long)tdmac); > + > + /* add the channel to tdma_chan list */ > + list_add_tail(&tdmac->chan.device_node, > + &tdev->device.channels); > + > + return 0; > +} > + > +static int __devinit mmp_tdma_probe(struct platform_device *pdev) > +{ > + const struct platform_device_id *id = platform_get_device_id(pdev); > + enum mmp_tdma_type type = id->driver_data; > + struct mmp_tdma_device *tdev; > + struct resource *iores; > + int i, ret; > + int irq = 0; > + int chan_num = TDMA_CHANNEL_NUM; > + > + /* always have couple channels */ > + tdev = devm_kzalloc(&pdev->dev, sizeof(*tdev), GFP_KERNEL); > + if (!tdev) > + return -ENOMEM; > + > + tdev->dev = &pdev->dev; > + iores = platform_get_resource(pdev, IORESOURCE_IRQ, 0); > + if (!iores) > + return -EINVAL; > + > + if (resource_size(iores) != chan_num) > + tdev->irq = iores->start; > + else > + irq = iores->start; > + > + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + if (!iores) > + return -EINVAL; > + > + tdev->base = devm_request_and_ioremap(&pdev->dev, iores); > + if (!tdev->base) > + return -EADDRNOTAVAIL; > + > + if (tdev->irq) { > + ret = devm_request_irq(&pdev->dev, tdev->irq, > + mmp_tdma_int_handler, IRQF_DISABLED, "tdma", tdev); > + if (ret) > + return ret; > + } > + > + dma_cap_set(DMA_SLAVE, tdev->device.cap_mask); > + dma_cap_set(DMA_CYCLIC, tdev->device.cap_mask); > + > + INIT_LIST_HEAD(&tdev->device.channels); > + > + /* initialize channel parameters */ > + for (i = 0; i < chan_num; i++) { > + ret = mmp_tdma_chan_init(tdev, i, irq, type); > + if (ret) > + return ret; > + } > + > + tdev->device.dev = &pdev->dev; > + tdev->device.device_alloc_chan_resources = > + mmp_tdma_alloc_chan_resources; > + tdev->device.device_free_chan_resources = > + mmp_tdma_free_chan_resources; > + tdev->device.device_prep_slave_sg = mmp_tdma_prep_slave_sg; > + tdev->device.device_prep_dma_cyclic = mmp_tdma_prep_dma_cyclic; > + tdev->device.device_tx_status = mmp_tdma_tx_status; > + tdev->device.device_issue_pending = mmp_tdma_issue_pending; > + tdev->device.device_control = mmp_tdma_control; > + tdev->device.copy_align = TDMA_ALIGNMENT; > + > + dma_set_mask(&pdev->dev, DMA_BIT_MASK(64)); > + platform_set_drvdata(pdev, tdev); > + > + ret = dma_async_device_register(&tdev->device); > + if (ret) { > + dev_err(tdev->device.dev, "unable to register\n"); > + return ret; > + } > + > + dev_info(tdev->device.dev, "initialized\n"); > + return 0; > +} > + > +static const struct platform_device_id mmp_tdma_id_table[] = { > + { "mmp-adma", MMP_AUD_TDMA }, > + { "pxa910-squ", PXA910_SQU }, > + { }, > +}; > + > +static struct platform_driver mmp_tdma_driver = { > + .driver = { > + .name = "mmp-tdma", > + .owner = THIS_MODULE, > + }, > + .id_table = mmp_tdma_id_table, > + .probe = mmp_tdma_probe, > + .remove = __devexit_p(mmp_tdma_remove), > +}; > + > +module_platform_driver(mmp_tdma_driver); > + > +MODULE_DESCRIPTION("MMP Two-Channel DMA Driver"); > +MODULE_LICENSE("GPL"); AUTHOR, ALIAS too pls > diff --git a/include/linux/platform_data/mmp_dma.h b/include/linux/platform_data/mmp_dma.h > new file mode 100644 > index 0000000..4e21cf9 > --- /dev/null > +++ b/include/linux/platform_data/mmp_dma.h > @@ -0,0 +1,20 @@ > +/* > + * MMP Platform DMA Management > + * > + * Copyright (c) 2011 Marvell Semiconductors Inc. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License version 2 as > + * published by the Free Software Foundation. > + * > + */ > + > +#ifndef MMP_DMA_H > +#define MMP_DMA_H > + > +struct mmp_tdma_data { > + u32 bus_size; > + u32 pack_mod; > +}; > + > +#endif /* MMP_DMA_H */ why do you need separate header for this? -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

Re: [alsa-devel] [PATCH v1 2/5] ASoC: mmp: add audio dma support
by Vinod Koul 07 Jun '12

07 Jun '12

On Mon, 2012-06-04 at 14:37 +0800, Zhangfei Gao wrote: > mmp-pcm handle audio dma based on soc-dmaengine > Support mmp and pxa910 See soc-dmaengine.c. You sneed to use the library to use dmaengine. > > Signed-off-by: Zhangfei Gao <zhangfei.gao(a)marvell.com> > Signed-off-by: Leo Yan <leoy(a)marvell.com> > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> > --- > include/linux/platform_data/mmp_audio.h | 22 ++ > sound/soc/pxa/Kconfig | 9 + > sound/soc/pxa/Makefile | 2 + > sound/soc/pxa/mmp-pcm.c | 317 +++++++++++++++++++++++++++++++ > 4 files changed, 350 insertions(+), 0 deletions(-) > create mode 100644 include/linux/platform_data/mmp_audio.h > create mode 100644 sound/soc/pxa/mmp-pcm.c > > diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h > new file mode 100644 > index 0000000..0f25d16 > --- /dev/null > +++ b/include/linux/platform_data/mmp_audio.h > @@ -0,0 +1,22 @@ > +/* > + * MMP Platform AUDIO Management > + * > + * Copyright (c) 2011 Marvell Semiconductors Inc. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License version 2 as > + * published by the Free Software Foundation. > + * > + */ > + > +#ifndef MMP_AUDIO_H > +#define MMP_AUDIO_H > + > +struct mmp_audio_platdata { > + u32 period_max_capture; > + u32 buffer_max_capture; > + u32 period_max_playback; > + u32 buffer_max_playback; > +}; > + > +#endif /* MMP_AUDIO_H */ > diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig > index a0f7d3c..5d76e29 100644 > --- a/sound/soc/pxa/Kconfig > +++ b/sound/soc/pxa/Kconfig > @@ -8,6 +8,15 @@ config SND_PXA2XX_SOC > the PXA2xx AC97, I2S or SSP interface. You will also need > to select the audio interfaces to support below. > > +config SND_MMP_SOC > + bool "Soc Audio for Marvell MMP chips" > + depends on ARCH_MMP > + select SND_SOC_DMAENGINE_PCM > + select SND_ARM > + help > + Say Y if you want to add support for codecs attached to > + the MMP SSPA interface. > + > config SND_PXA2XX_AC97 > tristate > select SND_AC97_CODEC > diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile > index af35762..f913e9b 100644 > --- a/sound/soc/pxa/Makefile > +++ b/sound/soc/pxa/Makefile > @@ -3,11 +3,13 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o > snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o > snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o > snd-soc-pxa-ssp-objs := pxa-ssp.o > +snd-soc-mmp-objs := mmp-pcm.o > > obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o > obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o > obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o > obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o > +obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o > > # PXA Machine Support > snd-soc-corgi-objs := corgi.o > diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c > new file mode 100644 > index 0000000..b7b09f9 > --- /dev/null > +++ b/sound/soc/pxa/mmp-pcm.c > @@ -0,0 +1,317 @@ > +/* > + * linux/sound/soc/pxa/mmp-pcm.c > + * > + * Copyright (C) 2011 Marvell International Ltd. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; either version 2 of the License, or > + * (at your option) any later version. > + * > + */ > +#include <linux/module.h> > +#include <linux/init.h> > +#include <linux/platform_device.h> > +#include <linux/slab.h> > +#include <linux/dma-mapping.h> > +#include <linux/dmaengine.h> > +#include <linux/platform_data/mmp_dma.h> > +#include <linux/platform_data/mmp_audio.h> > +#include <sound/pxa2xx-lib.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > +#include <mach/sram.h> > +#include <sound/dmaengine_pcm.h> > + > +struct mmp_dma_data { > + int ssp_id; > + struct resource *dma_res; > + struct mmp_tdma_data tdma_data; > +}; > + > +#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \ > + SNDRV_PCM_INFO_MMAP_VALID | \ > + SNDRV_PCM_INFO_INTERLEAVED | \ > + SNDRV_PCM_INFO_PAUSE | \ > + SNDRV_PCM_INFO_RESUME) > + > +#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ > + SNDRV_PCM_FMTBIT_S24_LE | \ > + SNDRV_PCM_FMTBIT_S32_LE) > + > +static struct snd_pcm_hardware mmp_pcm_hardware[] = { > + { > + .info = MMP_PCM_INFO, > + .formats = MMP_PCM_FORMATS, > + .period_bytes_min = 1024, > + .period_bytes_max = 2048, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 4096, > + .fifo_size = 32, > + }, > + { > + .info = MMP_PCM_INFO, > + .formats = MMP_PCM_FORMATS, > + .period_bytes_min = 1024, > + .period_bytes_max = 2048, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 4096, > + .fifo_size = 32, > + }, > +}; > + > +static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); > + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct pxa2xx_pcm_dma_params *dma_params; > + int ret; > + struct dma_slave_config slave_config; > + > + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); > + if (!dma_params) > + return 0; > + > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S8: > + dma_data->tdma_data.bus_size = 8; > + break; > + case SNDRV_PCM_FORMAT_S16_LE: > + dma_data->tdma_data.bus_size = 16; > + break; > + case SNDRV_PCM_FORMAT_S24_LE: > + dma_data->tdma_data.bus_size = 24; > + break; > + case SNDRV_PCM_FORMAT_S32_LE: > + dma_data->tdma_data.bus_size = 32; > + break; > + default: > + return -EINVAL; > + } > + dma_data->tdma_data.pack_mod = true; > + > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > + slave_config.direction = DMA_TO_DEVICE; > + slave_config.dst_addr = dma_params->dev_addr; > + slave_config.dst_maxburst = 4; > + } else { > + slave_config.direction = DMA_FROM_DEVICE; > + slave_config.src_addr = dma_params->dev_addr; > + slave_config.src_maxburst = 4; > + } > + > + ret = dmaengine_slave_config(chan, &slave_config); > + if (ret) > + return ret; > + > + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); > + > + return 0; > +} > + > +static bool filter(struct dma_chan *chan, void *param) > +{ > + struct mmp_dma_data *dma_data = param; > + bool found = false; > + char *devname; > + > + devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, > + dma_data->ssp_id); > + if ((strcmp(dev_name(chan->device->dev), devname) == 0) && > + (chan->chan_id == dma_data->dma_res->start)) { > + chan->private = &dma_data->tdma_data; > + found = true; > + } > + > + kfree(devname); > + return found; > +} > + > +static int mmp_pcm_open(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct platform_device *pdev = to_platform_device(rtd->platform->dev); > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + struct mmp_dma_data *dma_data; > + struct resource *r; > + int ret; > + > + r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); > + if (!r) > + return -EBUSY; > + > + snd_soc_set_runtime_hwparams(substream, > + &mmp_pcm_hardware[substream->stream]); > + dma_data = devm_kzalloc(&pdev->dev, > + sizeof(struct mmp_dma_data), GFP_KERNEL); > + if (dma_data == NULL) > + return -ENOMEM; > + > + dma_data->dma_res = r; > + dma_data->ssp_id = cpu_dai->id; > + > + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); > + if (ret) { > + devm_kfree(&pdev->dev, dma_data); > + return ret; > + } > + > + snd_dmaengine_pcm_set_data(substream, dma_data); > + return 0; > +} > + > +static int mmp_pcm_close(struct snd_pcm_substream *substream) > +{ > + struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct platform_device *pdev = to_platform_device(rtd->platform->dev); > + > + snd_dmaengine_pcm_close(substream); > + devm_kfree(&pdev->dev, dma_data); > + return 0; > +} > + > +static int mmp_pcm_mmap(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + unsigned long off = vma->vm_pgoff; > + > + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); > + return remap_pfn_range(vma, vma->vm_start, > + __phys_to_pfn(runtime->dma_addr) + off, > + vma->vm_end - vma->vm_start, vma->vm_page_prot); > +} > + > +struct snd_pcm_ops mmp_pcm_ops = { > + .open = mmp_pcm_open, > + .close = mmp_pcm_close, > + .ioctl = snd_pcm_lib_ioctl, > + .hw_params = mmp_pcm_hw_params, > + .trigger = snd_dmaengine_pcm_trigger, > + .pointer = snd_dmaengine_pcm_pointer, > + .mmap = mmp_pcm_mmap, > +}; > + > +static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) > +{ > + struct snd_pcm_substream *substream; > + struct snd_dma_buffer *buf; > + int stream; > + struct gen_pool *gpool; > + > + gpool = sram_get_gpool("asram"); > + if (!gpool) > + return; > + > + for (stream = 0; stream < 2; stream++) { > + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; > + > + substream = pcm->streams[stream].substream; > + if (!substream) > + continue; > + > + buf = &substream->dma_buffer; > + if (!buf->area) > + continue; > + gen_pool_free(gpool, (unsigned long)buf->area, size); > + buf->area = NULL; > + } > + > + return; > +} > + > +static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, > + int stream) > +{ > + struct snd_dma_buffer *buf = &substream->dma_buffer; > + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; > + struct gen_pool *gpool; > + > + buf->dev.type = SNDRV_DMA_TYPE_DEV; > + buf->dev.dev = substream->pcm->card->dev; > + buf->private_data = NULL; > + > + gpool = sram_get_gpool("asram"); > + if (!gpool) > + return -ENOMEM; > + > + buf->area = (unsigned char *)gen_pool_alloc(gpool, size); > + if (!buf->area) > + return -ENOMEM; > + buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area); > + buf->bytes = size; > + return 0; > +} > + > +int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_pcm_substream *substream; > + struct snd_pcm *pcm = rtd->pcm; > + int ret = 0, stream; > + > + for (stream = 0; stream < 2; stream++) { > + substream = pcm->streams[stream].substream; > + > + ret = mmp_pcm_preallocate_dma_buffer(substream, stream); > + if (ret) > + goto err; > + } > + > + return 0; > + > +err: > + mmp_pcm_free_dma_buffers(pcm); > + return ret; > +} > + > +struct snd_soc_platform_driver mmp_soc_platform = { > + .ops = &mmp_pcm_ops, > + .pcm_new = mmp_pcm_new, > + .pcm_free = mmp_pcm_free_dma_buffers, > +}; > + > +static __devinit int mmp_pcm_probe(struct platform_device *pdev) > +{ > + struct mmp_audio_platdata *pdata = pdev->dev.platform_data; > + > + if (pdata) { > + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max = > + pdata->buffer_max_playback; > + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max = > + pdata->period_max_playback; > + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max = > + pdata->buffer_max_capture; > + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max = > + pdata->period_max_capture; > + } > + return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform); > +} > + > +static int __devexit mmp_pcm_remove(struct platform_device *pdev) > +{ > + snd_soc_unregister_platform(&pdev->dev); > + return 0; > +} > + > +static struct platform_driver mmp_pcm_driver = { > + .driver = { > + .name = "mmp-pcm-audio", > + .owner = THIS_MODULE, > + }, > + > + .probe = mmp_pcm_probe, > + .remove = __devexit_p(mmp_pcm_remove), > +}; > + > +module_platform_driver(mmp_pcm_driver); > + > +MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); > +MODULE_DESCRIPTION("MMP Soc Audio DMA module"); > +MODULE_LICENSE("GPL"); -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 5

[alsa-devel] [PATCH] ALSA: hda - power setting error check
by Wang Xingchao 07 Jun '12

07 Jun '12

codec may reject power state transition requests(reporting PS-ERROR set), in that case we re-issue a power state setting and check error bit again. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d0ca370..bb64eba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3526,6 +3526,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { + int state, count = 0; if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3537,9 +3538,19 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); msleep(epss? 10:100); } +again: snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (state & AC_PWRST_ERROR) { + /* repeat power states setting at most 10 times*/ + if (count++ > 10) + snd_printk(KERN_WARNING "Power states transition reject!\n"); + else + goto again; + } } /* modem wake on ring: transition from D2 to D0 in less than 2ms. For modems, -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ALSA: hda - modem function group wake on ring support
by Wang Xingchao 07 Jun '12

07 Jun '12

D2 allows for the lowest possible power consuming state from which it can return to "fully on" state D0 within 2ms. For modems, it's the "wake on ring" power state. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2737fe7..d0ca370 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3542,6 +3542,18 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } +/* modem wake on ring: transition from D2 to D0 in less than 2ms. For modems, + * this is the wake-on-ring power state */ +static void hda_modem_wake_on_ring(struct hda_codec *codec, bool enter) +{ + if (codec->mfg) { + if (enter) + hda_set_power_state(codec, codec->mfg, AC_PWRST_D2); + else + hda_set_power_state(codec, codec->mfg, AC_PWRST_D0); + } +} + #ifdef CONFIG_SND_HDA_HWDEP /* execute additional init verbs */ static void hda_exec_init_verbs(struct hda_codec *codec) -- 1.7.9.5

3 3

Re: [alsa-devel] [PATCH] ALSA: hda - HDMI Audio init all connectors
by Takashi Iwai 07 Jun '12

07 Jun '12

At Thu, 07 Jun 2012 10:00:24 +0100, Steven Newbury wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/06/12 09:47, Takashi Iwai wrote: > > At the next time you submit a patch, please add Cc to maintainer > > and subsystem ML. Otherwise it'd be easily overlooked. > > > I meant to, somehow I managed to send it off before adding the CC, > probably too early in the morning! > > > At Thu, 07 Jun 2012 06:52:59 +0100, Steven Newbury wrote: > >> > >> When VGA_SWITCHEROO support is enabled hda_intel initialises the > >> HDMI audio device on the current VGA device. When it's not > >> enabled it only initialises the HDMI device on the default VGA > >> adaptor, this means secondary cards get no audio support which is > >> very unhelpful for multi-seat! > >> > >> With this patch, when SUPPORT_VGA_SWITCHEROO is disabled > >> hda_intel initialises all HDMI audio devices, not just the > >> default VGA. > >> > >> Signed-off-by: Steven Newbury <steve(a)snewbury.org.uk> > > > > OK, it must be fixed indeed. The bug was introduced due to the > > change in VGA-switcheroo side -- originally check_hdmi_disabled() > > really checked the VGA-switcheroo state, so it returned the correct > > value in the earlier version. > > > > In anyway, your patch looks almost OK, but it'd be better to move > > ifdef and simplify like below. Could you check whether it works > > for you? > > > Looks better, saves the pointless function call. I'll give it spin on > my next boot. Thanks! Meanwhile I found that the current version is also buggy for some setups with VGA-switcheroo. The current version won't work, for example, when VGA-switcheroo is built-in for systems without switcheroo. I'll need to cook up the fix for 3.5-rc2 in addition to your fix. Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - check proper return value
by Wang Xingchao 07 Jun '12

07 Jun '12

snd_hda_param_read() return value -1 means error, others are responses Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 79d0188..99316a7 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -451,7 +451,7 @@ static void print_power_state(struct snd_info_buffer *buffer, int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE); int pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); - if (sup) + if (sup != -1) snd_iprintf(buffer, " Power states: %s\n", bits_names(sup, names, ARRAY_SIZE(names))); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - check supported power states
by Wang Xingchao 07 Jun '12

07 Jun '12

Add function to check whether power states supported by specific codec node. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 41ca803..b89c8ec 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3505,6 +3505,22 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); /* + * supported power states check + */ +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); + + if (sup < 0) + return false; + if (sup & power_state) + return true; + else + return false; +} + +/* * set power state of the codec */ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, -- 1.7.9.5

4 4

[alsa-devel] [PATCH V2] ALSA: hda - reduce msleep time if EPSS power states supported
by Wang Xingchao 07 Jun '12

07 Jun '12

if EPSS supported, transition from D3 state to D0 state in less than 10ms Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b89c8ec..2737fe7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3532,8 +3532,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, } /* this delay seems necessary to avoid click noise at power-down */ - if (power_state == AC_PWRST_D3) - msleep(100); + if (power_state == AC_PWRST_D3) { + /* transition time less than 10ms for power down */ + bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); + msleep(epss ? 10 : 100); + } snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); -- 1.7.9.5

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - HDMI Audio init all connectors
by Takashi Iwai 07 Jun '12

07 Jun '12

At the next time you submit a patch, please add Cc to maintainer and subsystem ML. Otherwise it'd be easily overlooked. At Thu, 07 Jun 2012 06:52:59 +0100, Steven Newbury wrote: > > When VGA_SWITCHEROO support is enabled hda_intel initialises the HDMI audio device on the current VGA device. When it's not enabled it only initialises the HDMI device on the default VGA adaptor, this means secondary cards get no audio support which is very unhelpful for multi-seat! > > With this patch, when SUPPORT_VGA_SWITCHEROO is disabled hda_intel initialises all HDMI audio devices, not just the default VGA. > > Signed-off-by: Steven Newbury <steve(a)snewbury.org.uk> OK, it must be fixed indeed. The bug was introduced due to the change in VGA-switcheroo side -- originally check_hdmi_disabled() really checked the VGA-switcheroo state, so it returned the correct value in the earlier version. In anyway, your patch looks almost OK, but it'd be better to move ifdef and simplify like below. Could you check whether it works for you? Also, please give a patch that is applicable directly via git-am. Apparently you created it via git show. Instead, use git-format-patch or use --pretty=email for git-show. Or, send the patch directly via git-send-email. thanks, Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6392b..2eb1b47 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2484,9 +2484,9 @@ static void azx_notifier_unregister(struct azx *chip) static int DELAYED_INIT_MARK azx_first_init(struct azx *chip); static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip); +#ifdef SUPPORT_VGA_SWITCHEROO static struct pci_dev __devinit *get_bound_vga(struct pci_dev *pci); -#ifdef SUPPORT_VGA_SWITCHEROO static void azx_vs_set_state(struct pci_dev *pci, enum vga_switcheroo_state state) { @@ -2578,6 +2578,7 @@ static int __devinit register_vga_switcheroo(struct azx *chip) #else #define init_vga_switcheroo(chip) /* NOP */ #define register_vga_switcheroo(chip) 0 +#define check_hdmi_disabled(pci) false #endif /* SUPPORT_VGA_SWITCHER */ /* @@ -2638,6 +2639,7 @@ static int azx_dev_free(struct snd_device *device) return azx_free(device->device_data); } +#ifdef SUPPORT_VGA_SWITCHEROO /* * Check of disabled HDMI controller by vga-switcheroo */ @@ -2676,6 +2678,7 @@ static bool __devinit check_hdmi_disabled(struct pci_dev *pci) } return vga_inactive; } +#endif /* SUPPORT_VGA_SWITCHEROO */ /* * white/black-listing for position_fix

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Re: [alsa-devel] 3.5-rc1 snd_usb_audio stream format breakage
by Takashi Iwai 07 Jun '12

07 Jun '12

At Thu, 7 Jun 2012 03:38:30 +0000, Jamie Heilman wrote: > > With the USB audio streaming logic changes, my HeadRoom Bithead > (08bb:2902 Burr-Brown PCM2902) no longer works. Symptoms include the > loss of most of the normally available formats leaving only: > > root@cucamonga:~# cat /proc/asound/bithead/stream0 > Burr-Brown from TI USB Audio CODEC at usb-0000:00:1a.7-1.1.4, full speed : USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 6 > Format: U8 > Channels: 1 > Endpoint: 2 OUT (ADAPTIVE) > Rates: 32000, 44100, 48000 > > Capture: > Status: Stop > Interface 2 > Altset 18 > Format: S8 > Channels: 1 > Endpoint: 4 IN (SYNC) > Rates: 11025 > > (normally 2 channel S16_LE is available, with capture rates up to > 48kHz ... only the last altset's are being reported now) and a slew of > proc registration warnings logged: > > ------------[ cut here ]------------ > WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6() > Hardware name: Precision WorkStation T3400 > proc_dir_entry 'card1/pcm0p' already registered > Modules linked in: snd_usb_audio(+) snd_usbmidi_lib snd_hda_intel(+) snd_hda_codec snd_hwdep snd_seq_midi snd_seq_midi_event snd_pcm snd_page_alloc snd_rawmidi snd_seq snd_seq_device snd_timer snd soundcore dcdbas floppy sr_mod usb_storage cdrom kvm_intel psmouse kvm tg3 evdev xfs dm_mod raid1 md_mod > Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2 > Call Trace: > [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c > [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48 > [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b > [<ffffffff811292f0>] proc_register+0x169/0x1a6 > [<ffffffff8112962e>] create_proc_entry+0x74/0x8c > [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd] > [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm] > [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio] > [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio] > [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb > [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio] > [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio] > [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio] > [<ffffffff812c49f5>] usb_probe_interface+0x136/0x1b1 > [<ffffffff81287e66>] ? driver_probe_device+0x1d3/0x1d3 > [<ffffffff81287d43>] driver_probe_device+0xb0/0x1d3 > [<ffffffff81287ec5>] __driver_attach+0x5f/0x83 > [<ffffffff812863a2>] bus_for_each_dev+0x54/0x86 > [<ffffffff81287730>] driver_attach+0x1e/0x20 > [<ffffffff8128747f>] bus_add_driver+0xfd/0x23a > [<ffffffff812882e8>] driver_register+0x93/0x10e > [<ffffffff812c383f>] usb_register_driver+0x7e/0x12a > [<ffffffffa0257000>] ? 0xffffffffa0256fff > [<ffffffffa0257000>] ? 0xffffffffa0256fff > [<ffffffffa0257040>] snd_usb_audio_init+0x40/0x1000 [snd_usb_audio] > [<ffffffff810001ff>] do_one_initcall+0x7f/0x134 > [<ffffffffa0257000>] ? 0xffffffffa0256fff > [<ffffffff810688d4>] sys_init_module+0x16cf/0x195e > [<ffffffff8106516c>] ? free_notes_attrs+0x4d/0x4d > [<ffffffff810c38cb>] ? mmap_region+0x1c7/0x469 > [<ffffffff813856d2>] system_call_fastpath+0x16/0x1b > ---[ end trace c1baea1a25a91564 ]--- Looks like the PCM substream assignment is buggy. Does the patch (untested at all) below help? thanks, Takashi --- diff --git a/sound/usb/card.h b/sound/usb/card.h index 0d37238..2b9ffff 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -119,6 +119,7 @@ struct snd_usb_substream { unsigned long unlink_mask; /* bitmask of unlinked urbs */ /* data and sync endpoints for this stream */ + unsigned int ep_num; /* the endpoint number */ struct snd_usb_endpoint *data_endpoint; struct snd_usb_endpoint *sync_endpoint; unsigned long flags; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 6b7d7a2..083ed81 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -97,6 +97,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->formats |= fp->formats; subs->num_formats++; subs->fmt_type = fp->fmt_type; + subs->ep_num = fp->endpoint; } /* @@ -119,9 +120,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (!subs->data_endpoint) - continue; - if (subs->data_endpoint->ep_num == fp->endpoint) { + if (subs->ep_num == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; subs->formats |= fp->formats; @@ -134,7 +133,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (subs->data_endpoint) + if (subs->ep_num) continue; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0)

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[alsa-devel] [PATCH V2] ALSA: hda - add power states information in proc
by Wang Xingchao 06 Jun '12

06 Jun '12

add more power states information: - reset status - clock stop ok - power states error Output like: Power: setting=D0, actual=D0, Error, Clock-stop-OK, Setting-reset Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 9 ++++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960..71864cdd 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -323,6 +323,9 @@ enum { #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 #define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e59e2f0..79d0188 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -455,10 +455,17 @@ static void print_power_state(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Power states: %s\n", bits_names(sup, names, ARRAY_SIZE(names))); - snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + snd_iprintf(buffer, " Power: setting=%s, actual=%s", get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT)); + if (pwr & AC_PWRST_ERROR) + snd_iprintf(buffer, " , Error"); + if (pwr & AC_PWRST_CLK_STOP_OK) + snd_iprintf(buffer, " , Clock-stop-OK"); + if (pwr & AC_PWRST_SETTING_RESET) + snd_iprintf(buffer, " , Setting-reset"); + snd_iprintf(buffer, "\n"); } static void print_unsol_cap(struct snd_info_buffer *buffer, -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH v2 2/2] ASoC: codecs: Add DA732x codec driver
by Adam Thomson 06 Jun '12

06 Jun '12

On Wed, June 4, 2012 at 15:31, Mark Brown wrote: > A few small things below, very minor and easy to correct though. > > > + { DA732X_REG_STATUS_EXT , 0x00 }, > > + { DA732X_REG_STATUS , 0x00 }, > > > + { DA732X_REG_MBOX0 , 0x00 }, > > + { DA732X_REG_MBOX1 , 0x00 }, > > + { DA732X_REG_MBOX2 , 0x00 }, > > + { DA732X_REG_MBOX_STATUS , 0x00 }, > > > + { DA732X_REG_ID , 0x00 }, > > > + { DA732X_REG_DMA_STATUS , 0x00 }, > > + { DA732X_REG_BROWNOUT , 0x00 }, > > I would be very surprised if some or all of the above regsters aren't > volatile and therefore shouldn't have default specified. Correct. Will remove those from the cache. > > +static int da732x_hpf_set(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) > > Why doesn't a standard enum work for this? This is because the two bits that are required for this are not adjacent in the register, so we use our own get/set methods to control them. > > + /* High Pass Filters */ > > + SOC_ENUM_EXT("DAC1 High Pass Filter Mode Switch", > > + da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), > > This shouldn't be a Switch, it's an enumeration. Switches are plain > boolean controls. Ok, fair point. Will rename accordingly. > > + SOC_SINGLE("ADC1 Equalizer Switch", DA732X_REG_ADC1_EQ5, > > + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), > > + SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12, > > + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, > > + DA732X_INVERT, eq_band_pga_tlv), > > I'd suggest calling the Switch "ADC1 EQ Switch" for consistency with the > gains for the bands (or expanding EQ in the gains but that'll make for a > lot of really long control names). Again, makes sense. Will update all the ADC & DAC instances to align. > > + case SND_SOC_DAPM_POST_PMU: > > + if (w->reg == DA732X_REG_ADC1_PD) > > + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, > > + DA732X_ADCA_BB_CLK_EN, > > + DA732X_ADCA_BB_CLK_EN); > > + else > > + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, > > + DA732X_ADCC_BB_CLK_EN, > > + DA732X_ADCC_BB_CLK_EN); > > This reads like it should be a switch statement. The effect will be the > same but it's a bit clearer. Ok, don't mind changing it if you think it's clearer. > > + da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap); > > + if (IS_ERR(da732x->regmap)) { > > + ret = PTR_ERR(da732x->regmap); > > + dev_err(&i2c->dev, "Failed to initialize regmap\n"); > > + goto err; > > + } > > + > > + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, > > + da732x_dai, ARRAY_SIZE(da732x_dai)); > > The register cache defaults make it look like there's a device ID > (or possibly chip revision?) register - it'd be good practice to read it > here and either verify that it has the expected value or log it > depending on which it is. Is revision information. Will log this. > > +static const struct i2c_device_id da732x_i2c_id[] = { > > + { "da732x", 0}, > > + { } > > +}; > > +MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); > > Good practice would be to enumerate all the device IDs (so users can > just register devices) but it doesn't really matter. For now will stick with this.

1 0

[alsa-devel] [PATCH] ASoC:isabelle:Remove regmap_exit()
by MR.Swami.Reddy＠ti.com 06 Jun '12

06 Jun '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> ASoC:isabelle:With devm_ APIs regmap_exit() not needed, so remove regmap_exit(). Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com> Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/isabelle.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 69228ec..bcc77ef 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1141,7 +1141,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(isabelle_dai)); if (ret < 0) { dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); - regmap_exit(dev_get_regmap(&i2c->dev, NULL)); return ret; } @@ -1151,7 +1150,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, static int __devexit isabelle_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - regmap_exit(dev_get_regmap(&client->dev, NULL)); return 0; }

3 2

[alsa-devel] [PATCH v3] ASoC: omap-abe-twl6040: Add device tree support
by Peter Ujfalusi 06 Jun '12

06 Jun '12

When the board boots with device tree the driver will receive the name of the card, DAPM routing map, phandle for the audio components described in the dts file, mclk speed, and the possibility of detecting the jack detection. The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reviwed-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Hi, Changes since v2 (resend): - Do not override the return value from snd_soc_of_parse_audio_routing in case of error. Regards, Peter .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 ++++++++++++ sound/soc/omap/omap-abe-twl6040.c | 145 ++++++++++++++++---- 2 files changed, 206 insertions(+), 30 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 0000000..65dec87 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack_detection: Need to be set to <1> if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection = <1>; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 9d93793..1343147 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -25,6 +25,7 @@ #include <linux/mfd/twl6040.h> #include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,6 +44,8 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ + + struct platform_device *dmic_codec_dev; }; static int omap_abe_hw_params(struct snd_pcm_substream *substream, @@ -185,17 +188,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) int hs_trim; int ret = 0; - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. @@ -216,6 +208,24 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } + /* + * NULL pdata means we booted with DT. In this case the routing is + * provided and the card is fully routed, no need to mark pins. + */ + if (!pdata) + return ret; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); + twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); + twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); + return ret; } @@ -270,52 +280,116 @@ static struct snd_soc_card omap_abe_card = { static __devinit int omap_abe_probe(struct platform_device *pdev) { struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; struct abe_twl6040 *priv; int num_links = 0; - int ret; + int ret = 0; card->dev = &pdev->dev; - if (!pdata) { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; - } - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (pdata->card_name) { - card->name = pdata->card_name; + priv->dmic_codec_dev = ERR_PTR(-EINVAL); + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + ret = snd_soc_of_parse_audio_routing(card, + "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, + "Error while parsing DAPM routing\n"); + return ret; + } + + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_dai_of_node = dai_node; + + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_dai_of_node = dai_node; + + priv->dmic_codec_dev = platform_device_register_simple( + "dmic-codec", -1, NULL, 0); + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, + "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); + } + } else { + num_links = 1; + } + + of_property_read_u32(node, "ti,jack-detection", + &priv->jack_detection); + of_property_read_u32(node, "ti,mclk-freq", + &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; + } + + omap_abe_card.fully_routed = 1; + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + if (pdata->has_dmic) + num_links = 2; + else + num_links = 1; + + priv->jack_detection = pdata->jack_detection; + priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Card name is not provided\n"); + dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; - if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - return -ENODEV; + ret = -ENODEV; + goto err_unregister; } - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - card->dai_link = abe_twl6040_dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_unregister; + } + + return 0; + +err_unregister: + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -323,17 +397,28 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) static int __devexit omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); + return 0; } +static const struct of_device_id omap_abe_of_match[] = { + {.compatible = "ti,abe-twl6040", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_abe_of_match); + static struct platform_driver omap_abe_driver = { .driver = { .name = "omap-abe-twl6040", .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, .remove = __devexit_p(omap_abe_remove), -- 1.7.8.6

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[alsa-devel] [PATCH 9/8] printk: Only look for prefix levels in kernel messages
by Joe Perches 06 Jun '12

06 Jun '12

vprintk_emit prefix parsing should only be done for internal kernel messages. This allows existing behavior to be kept in all cases. Signed-off-by: Joe Perches <joe(a)perches.com> --- kernel/printk.c | 32 +++++++++++++++++--------------- 1 files changed, 17 insertions(+), 15 deletions(-) diff --git a/kernel/printk.c b/kernel/printk.c index 5cd73f7..4e72c07 100644 --- a/kernel/printk.c +++ b/kernel/printk.c @@ -1267,7 +1267,6 @@ asmlinkage int vprintk_emit(int facility, int level, static char cont_buf[LOG_LINE_MAX]; static size_t cont_len; static int cont_level; - int kern_level; static struct task_struct *cont_task; static char textbuf[LOG_LINE_MAX]; char *text = textbuf; @@ -1329,21 +1328,24 @@ asmlinkage int vprintk_emit(int facility, int level, newline = true; } - /* strip syslog prefix and extract log level or control flags */ - kern_level = printk_get_level(text); - if (kern_level) { - const char *end_of_header = printk_skip_level(text); - switch (kern_level) { - case '0' ... '7': - if (level == -1) - level = kern_level - '0'; - case 'd': /* KERN_DEFAULT */ - prefix = true; - case 'c': /* KERN_CONT */ - break; + /* strip kernel syslog prefix and extract log level or control flags */ + if (facility == 0) { + int kern_level = printk_get_level(text); + + if (kern_level) { + const char *end_of_header = printk_skip_level(text); + switch (kern_level) { + case '0' ... '7': + if (level == -1) + level = kern_level - '0'; + case 'd': /* KERN_DEFAULT */ + prefix = true; + case 'c': /* KERN_CONT */ + break; + } + text_len -= end_of_header - text; + text = (char *)end_of_header; } - text_len -= end_of_header - text; - text = (char *)end_of_header; } if (level == -1)

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 03:10 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 2:46 AM, Andrew Morton <akpm(a)linux-foundation.org> wrote: > > On Tue, 05 Jun 2012 17:40:05 -0700 Joe Perches <joe(a)perches.com> wrote: > > > >> On Tue, 2012-06-05 at 17:37 -0700, Andrew Morton wrote: > >> > On Tue, 05 Jun 2012 17:07:27 -0700 Joe Perches <joe(a)perches.com> wrote: > >> > > >> > > On Tue, 2012-06-05 at 16:58 -0700, Andrew Morton wrote: > >> > > > echo "\0014Hello Joe" > /dev/kmsg > >> > > > >> > > # echo -e "\x014Hello Me" > /dev/kmsg > >> > > gives: > >> > > 12,778,4057982669;Hello Me > >> > > >> > That's changed behavior. > >> > >> Which is an improvement too. > > > > No it isn't. It exposes internal kernel implementation details in > > random weird inexplicable ways. It doesn't seem at all important > > though. > > > >> I very much doubt a single app will change > >> because of this. > > > > I doubt it as well. > > Yeah, the value of injecting such binary data is kind of questionable. :) > > Joe, maybe you can change printk_emit() to skip the prefix > detection/stripping if a prefix is already passed to the function? Sure, it's pretty trivial. Perhaps all binary data data should be elided. Maybe print . for all non-printable ascii chars.

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Andrew Morton 06 Jun '12

06 Jun '12

On Tue, 05 Jun 2012 17:07:27 -0700 Joe Perches <joe(a)perches.com> wrote: > On Tue, 2012-06-05 at 16:58 -0700, Andrew Morton wrote: > > echo "\0014Hello Joe" > /dev/kmsg > > # echo -e "\x014Hello Me" > /dev/kmsg > gives: > 12,778,4057982669;Hello Me That's changed behavior. On Wed, 6 Jun 2012 02:28:39 +0200 Kay Sievers <kay(a)vrfy.org> wrote: > Yeah, but printk_emit() will not try to parse it? I did not check, but > with your change, the prefix parsing in printk_emit() is still skipped > if a level is given as a parameter to printk_emit(), right? printk_emit() does parse the leading \0014, and then skips over it, removing it from the output stream. printk_emit() then throws away the resulting level because devkmsg_writev() did not pass in level==-1.

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 02:28 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 2:19 AM, Joe Perches <joe(a)perches.com> wrote: > > On Wed, 2012-06-06 at 02:13 +0200, Kay Sievers wrote: > >> The question is what happens if you inject your new binary two-byte > >> prefix, like: > >> echo -e "\x01\x02Hello" > /dev/kmsg > > > > It's not a 2 byte binary. > > It's a leading ascii SOH and a standard ascii char > > '0' ... '7' or 'd'. > > > > #define KERN_EMERG KERN_SOH "0" /* system is unusable */ > > #define KERN_ALERT KERN_SOH "1" /* action must be taken immediately */ > > etc... > > Ok. > > >> And if that changes the log-level to "2" instead of the default "4"? > > > > No it doesn't. > > So: > echo -e "\x012Hello" > /dev/kmsg > is still level 4? Sounds all fine then. Yes. # echo -e "\x012Hello again Kay" > /dev/kmsg gives: 12,780,6031964979;Hello again Kay > > It's not triggering that because devkmsg_writev does > > prefix parsing only on the old "<n>" form. > > Yeah, but printk_emit() will not try to parse it? I did not check, but > with your change, the prefix parsing in printk_emit() is still skipped > if a level is given as a parameter to printk_emit(), right? If level is not -1, then whatever prefix level the string has is ignored by vprintk_emit. from vprintk_emit: /* strip syslog prefix and extract log level or control flags */ kern_level = printk_get_level(text); if (kern_level) { const char *end_of_header = printk_skip_level(text); switch (kern_level) { case '0' ... '7': if (level == -1) level = kern_level - '0'; case 'd': /* Strip d KERN_DEFAULT, start new line */ plen = 0; default: if (!new_text_line) { log_buf_emit_char('\n'); new_text_line = 1; } } text_len -= end_of_header - text; text = (char *)end_of_header; } Only level == -1 will use the prefix level. devkmsg_writev always passes a non -1 level. cheers, Joe

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 02:13 +0200, Kay Sievers wrote: > The question is what happens if you inject your new binary two-byte > prefix, like: > echo -e "\x01\x02Hello" > /dev/kmsg It's not a 2 byte binary. It's a leading ascii SOH and a standard ascii char '0' ... '7' or 'd'. #define KERN_EMERG KERN_SOH "0" /* system is unusable */ #define KERN_ALERT KERN_SOH "1" /* action must be taken immediately */ etc... > And if that changes the log-level to "2" instead of the default "4"? No it doesn't. It's not triggering that because devkmsg_writev does prefix parsing only on the old "<n>" form.

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 01:48 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 1:43 AM, Joe Perches <joe(a)perches.com> wrote: > > On Wed, 2012-06-06 at 01:39 +0200, Kay Sievers wrote: > > >> > # echo "\001Hello Andrew" > /dev/kmsg > >> > /dev/kmsg has > >> > 12,774,2462339252;\001Hello Andrew > >> > >> Try "echo -e"? The stuff is copied verbatim otherwise. > > > > # echo -e "\001Hello Kay" > /dev/kmsg > > gives > > 12,776,3046752764;\x01Hello Kay > > Don't you need two bytes to trigger the logic? Yes. Angle brackets fore and aft. If you use a "<n>" prefix for write to /dev/kmsg, then it's supported just like before. Otherwise, it's emitted at KERN_DEFAULT. cheers, Joe

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 01:39 +0200, Kay Sievers wrote: > Try "echo -e"? The stuff is copied verbatim otherwise. btw: I didn't change the devkmsg_writev function which does the decoding of any "<n>" prefix for printk_emit.

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