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[alsa-devel] ALSA Programming Query
by santosh pattar 23 Oct '08

23 Oct '08

Dear All, I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, santosh

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[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 23 Oct '08

23 Oct '08

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

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[alsa-devel] Fwd: multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Sean McNamara 22 Oct '08

22 Oct '08

Oops! Forgot to include the ML on this reply. Maybe someone else will find this info useful, or be able to add constructive comments to what has been said :) Sean ---------- Forwarded message ---------- From: Sean McNamara <smcnam(a)gmail.com> Date: Wed, Oct 22, 2008 at 4:17 PM Subject: Re: [alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters) To: Jelle de Jong <jelledejong(a)powercraft.nl> On Wed, Oct 22, 2008 at 1:48 PM, Jelle de Jong <jelledejong(a)powercraft.nl> wrote: > Hello everybody, > > I have a four to eighth muliseat debian linux system here and it is > missing sound for the users. I was hoping the alsa team can help me with > this issue. All sound issues on Linux are extremely application-specific. Getting sound to work without any specific application in mind is asking for trouble, because there is no single configuration of the sound stack (which, if you expand into historical sound APIs and servers, contains about 20 different ways to play sound) that will satisfy any application without further configuration. Since this is not a really well-formed question, I will proceed with the assumption that you want to be able to run, for example, aplay /usr/share/sounds/pop.wav in the console, and have it produce sound in the "appropriate" headset. Here's how you proceed: 1. Each person on the multiseat box must have their own UNIX user account. 2. Each user account must have an ~/.asoundrc file. 3. Each ~/.asoundrc file must redefine the "default" device. How you redefine it depends on a lot of things: (a) Do you want _direct_ hardware access to the sound card, i.e. one application at a time? (b) Do you want to use ALSA's built-in software mixing, i.e. dmix? (c) How many channels? (d) Do you want to use PulseAudio? (e) Do you want to use JACK? (f) For each USB headset, what is its corresponding "raw device string"? This will be something like hw:n, where n is a number starting from 0, depending on the order in which the USB drivers initialize each USB controller. All of the above questions, and others, will determine exactly how each user configures their ~/.asoundrc file. The *extremely naive* (not recommended) example of an ~/.asoundrc is like: pcm.!default { card 3 } to have ALSA clients play sound, by default, to the fourth sound device that happened to be initialized. > > I have usb headsets and usb audio devices and all users have there own > usb hub. There are no internal audio cards in the multiseat server. > > How can we configure an system that the user can listen to audio only to > the devices connected to his usb hub. Well, this is guaranteed by design, isn't it? Without physical access to someone else's headset, how would I get access to the audio streams being played to that headset? I don't think this really expresses what your problem is. If we flip your question around, it might be a little more interesting: "How can we configure a system so that each user's applications will only play audio to that user's corresponding headset?" You can interpret this question in two ways: 1. "How can we _prevent_ users from playing sound to each other's sound cards?" [mutually untrust[ed/ing] users] or 2. "How can we configure the default settings so that, if untampered with, a user's applications will play sound to that user's own headset and not anyone else's?" [mutually trust[ed/ing] users] The lack of hardware mixing actually comes back to being a security benefit to user space control in this kind of situation. Recall that if I run aplay --device=hw:0 /usr/share/sounds/pop.wav then as long as the aplay program is playing sound to hw:0, it is *impossible* (short of some really devious hacking in alsa-lib or the kernel) for another application -- from this user, or another -- to also play sound to hw:0 at the same time. I'll call this claim "No Native SW Mixing". Now, let's look at what is required to satisfy the question in each of the two above cases. Trusted: 1. We know that users aren't going to maliciously try and hurt one another's ears by setting someone else's mixer really loud and playing static to them. 2. We know that users won't try to tamper with eachother's processes or configuration files; once things are configured correctly, it'll stay that way. 3. By 'No Native SW Mixing', we know that users can *theoretically* hog another user's sound device whenever that user has zero applications currently using it. 4. Since users trust each other, they won't do this. 5. We can use something like dmix, or just use hw:0 if users don't need simultaneous app output, and things will be fine. 6. dmix is a viable option to implement software mixing. 7. PulseAudio is also a viable option to implement software mixing. Untrusted: 1. Users may want to maliciously try and hurt one another's ears by setting someone else's mixer really loud and playing static to them. 2. Users may try to tamper with eachother's processes or configuration files; once things are configured correctly, it could get messed up again. [There's not much you can do to prevent this ... at least not based on existing open source tools.] 3. By 'No Native SW Mixing', we know that users can *theoretically* hog another user's sound device whenever that user has zero applications currently using it. 4. Since users do not trust each other, there is potential for 3 to happen. 5. We must use 'No Native SW Mixing' to our advantage, to create a walled garden around each user's sound experience. We need a process that, once started by a user, will permanently "hog" the sound card that user wants to use. This process should, ideally, allow other processes run by that user (and by that user only) to access their USB headset for output or input. We need a "gatekeeper" that is sensitive to the context in which an app is being run. 6. dmix is a NOT viable option to implement software mixing, because: (a) Once all dmix applications close their streams, the audio device hw:n is available, and some malicious user can decide to acquire it, abuse it, and prevent its rightful user from accessing it with his own applications. (b) dmix does not restrict users from outputting sound based on their UID or session. It is NOT a gatekeeper, just an open gate. 7. PulseAudio is [perhaps the best] viable solution to implement software mixing for each respective user. It is also a viable solution to preventing users from taking control of one another's sound devices. Configuring pulseaudio for a multi-soundcard multi-user system is out of the scope of this mailing list, but suffice it to say that it can be done. You will want to run one PulseAudio daemon per user, and each daemon will hold exclusive control over exactly one USB headset. Each daemon will only accept clients' requests for audio I/O from the user who started the daemon. Do make sure that you disable module-suspend-on-idle, or PulseAudio will actually give up the sound device after a period of inactivity, making it insecure in an untrusted environment. BTW, if this project is for a commercial interest, you usually pay someone for this kind of in-depth analysis. If you're that someone, you might want to read up on some of the underlying technologies so that you can develop this sort of reasoning on your own. I can envision a very real situation for both the trusted and the untrusted environment, so there is definitely a demand for this kind of specialist. Trusted environments are popular at workplaces, and untrusted multi-seat boxes are popular at computer labs, public libraries, etc. > > Any help is really appreciated. Please let me know if this information was helpful -- it will help me gauge, in turn, whether I should spend my time writing up these sorts of answers on the ML. FWIW, I searched around for existing articles touching this subject, but I couldn't find anyone who addresses this specific issue accurately or in enough detail. Sean P.S. - I once spoke to a user on IRC who was having intermittent problems with multiple USB headsets. It turns out that the USB controllers in the computer didn't have enough bandwidth, or power (or both) to allow all of the headsets to play at once. If, after configuring, you run into issues where only so many users can play sound at once, then you should invest in additional PCI or (preferably) PCI-Express USB controllers. Note that adding hubs upstream of a USB controller does absolutely nothing to lessen the load on the USB controllers that are integrated into your motherboard. > > Thank in advance, > > Jelle > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Jelle de Jong 22 Oct '08

22 Oct '08

Hello everybody, I have a four to eighth muliseat debian linux system here and it is missing sound for the users. I was hoping the alsa team can help me with this issue. I have usb headsets and usb audio devices and all users have there own usb hub. There are no internal audio cards in the multiseat server. How can we configure an system that the user can listen to audio only to the devices connected to his usb hub. Any help is really appreciated. Thank in advance, Jelle

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 22 Oct '08

22 Oct '08

At Tue, 21 Oct 2008 14:46:19 +0200, Nir Tzachar wrote: > > >> The bad news are that mplayer misbehaves and will get stuck every time > >> I try to change the volume. However, this does not happen in vlc. Any > >> ideas? > > > > Well, we need to sort out where the problem comes from. > > Try the vanilla Linus tree without sound-2.6 whether the problem > > persists. > > I'll try to remove the sound-2.6 tree and report. Nir, do you have any results? I tried to reproduce the problem by myself, but I couldn't. So I wonder whether it's a codec specific or I missed something... thanks, Takashi

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[alsa-devel] [PATCH] ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver
by Jarkko Nikula 22 Oct '08

22 Oct '08

Fix "ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver" was not correct due misunderstanding of DSP_A format and similar error in TLV320AIC33 codec which was used to test the original fix. This patch corrects now DSP_A format in OMAP McBSP DAI driver and is verified with TLV320AIC23 codec that's implementing DSP_A correctly. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Arun KS <arunks(a)mistralsolutions.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 853b33a..8485a8a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A: regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(0); + regs->srgr1 |= FWID(wlen * 2 - 2); break; } @@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); - /* Invert bit clock and FS polarity configuration for DSP_A */ - temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- 1.5.6.5

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[alsa-devel] [PATCH 2.6.28] ALSA: Ensure PXA runtime data is initialised
by Mark Brown 22 Oct '08

22 Oct '08

The rest of the code relies on the runtime data being zero initialised so we need to use kzalloc() to allocate it. Reported-by: Oliver Ford <ipaqlinux(a)oliford.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb7..75a0d74 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = -- 1.5.6.5

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Re: [alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Tony Vroon 22 Oct '08

22 Oct '08

On a related note, it looks to me as if the leading two characters (the manufacturing location) have been erroneously included for E-MU products. You'll see this MA code on some sound cards as well (MASBxxxx) Regards, Tony V.

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[alsa-devel] [PATCH] ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver
by Jarkko Nikula 22 Oct '08

22 Oct '08

Fix word clock length which must equal to one bit clock cycle in DSP mode. Surprisingly McBSP is able synchronize into wrong length when it's slave but e.g. TLV320AIC33 codec in slave configuration is outputting some amount of noise if word clock length is longer than one bit clock cycle. Fix also bit clock and frame sync polarities in DSP mode since they are opposite from I2S. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Arun KS <arunks(a)mistralsolutions.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 24 ++++++++++++++++++++---- 1 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 0a063a9..853b33a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -43,6 +43,7 @@ struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; /* * Flags indicating is the bus already activated and configured by * another substream @@ -200,6 +201,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + int wlen; unsigned long port; if (cpu_class_is_omap1()) { @@ -244,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ + wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); - /* Set FS period and length in terms of bit clock periods */ - regs->srgr2 |= FPER(16 * 2 - 1); - regs->srgr1 |= FWID(16 - 1); break; default: /* Unsupported PCM format */ return -EINVAL; } + /* Set FS period and length in terms of bit clock periods */ + switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen - 1); + break; + case SND_SOC_DAIFMT_DSP_A: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(0); + break; + } + omap_mcbsp_config(bus_id, &mcbsp_data->regs); mcbsp_data->configured = 1; @@ -272,10 +284,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -293,6 +307,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert bit clock and FS polarity configuration for DSP_A */ + temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -316,7 +332,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Vedran Miletić 22 Oct '08

22 Oct '08

ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style Changelog: * added missing SBxxxx, CTxxxx, PCxxx and MAEMxxxx where they were missing, and fixed some of them which were wrong (according to kx.inf, which is pretty accurate compared to anything out there) * fixed device names to make them more consistent across various cards * fixed lots of coding style errors and some warnings, now it is down to: checkpatch.pl-0.24 --no-tree --file --strict --terse emu10k1_main.c total: 0 errors, 62 warnings, 7 checks, 2074 lines checked Signed-off-by: Vedran Miletic <rivanvx(a)gmail.com> -- Vedran Miletić

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