July 2012 - Alsa-devel - mailman.alsa-project.org

[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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