Re: [Alsa-user] Output latency
stan wrote: Florian Faber wrote: You want hardware monitoring - there are sound cards that support hardware mixing. With good converters you have latencies down to 5 samples at 192kHz, that would be 0.026ms for each way, 0.052ms over all. I'm not the original poster, but I'm curious about this. Can you name a card that has this capability supported in alsa? I know that CMI8788-based cards can do this. There may be other cards, like the ESI Julia, but I don't know if monitoring is supported by the driver. Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output latency
Alexander Carôt wrote: 3.) Rather than using a double buffer for the playout wouldn't it be possible to choose only one physical playout buffer and parse the captured data in right at the interrupt. It's unlikely that any code could be fast enough to write the entire buffer before the hardware starts reading from the buffer; in fact, the hardware is likely to read the first sample from the buffer at the same time it is triggering the interrupt for the previous period. To reduce output latency, just reduce the output buffer size. For example, you could use a buffer with 128 samples (with 2 64-sample periods). HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Audio hardware with pause support
Hi, The snd_pcm_pause function of the ALSA API is not supported on all audio hardware. Is there an official list (e.g. on the web) of known sound hardware, which supports this feature? Is there another way to determine whether a certain hardware supports snd_pcm_pause without having to test the hardware? Best regards, Florian - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Audio hardware with pause support
Florian Winter wrote: Is there another way to determine whether a certain hardware supports snd_pcm_pause without having to test the hardware? $ grep -rl SNDRV_PCM_INFO_PAUSE sound sound/arm/pxa2xx-pcm.c sound/arm/sa11xx-uda1341.c sound/core/pcm_native.c sound/drivers/vx/vx_pcm.c sound/pci/als300.c sound/pci/atiixp.c sound/pci/au88x0/au88x0_pcm.c sound/pci/cmipci.c sound/pci/cs4281.c sound/pci/cs5535audio/cs5535audio_pcm.c sound/pci/echoaudio/darla20.c sound/pci/echoaudio/darla24.c sound/pci/echoaudio/echo3g.c sound/pci/echoaudio/gina20.c sound/pci/echoaudio/gina24.c sound/pci/echoaudio/indigo.c sound/pci/echoaudio/indigodj.c sound/pci/echoaudio/indigoio.c sound/pci/echoaudio/layla20.c sound/pci/echoaudio/layla24.c sound/pci/echoaudio/mia.c sound/pci/echoaudio/mona.c sound/pci/emu10k1/emupcm.c sound/pci/ens1370.c sound/pci/es1968.c sound/pci/fm801.c sound/pci/hda/hda_intel.c sound/pci/ice1712/ice1712.c sound/pci/ice1712/ice1724.c sound/pci/intel8x0.c sound/pci/intel8x0m.c sound/pci/maestro3.c sound/pci/mixart/mixart.c sound/pci/nm256/nm256.c sound/pci/oxygen/oxygen_pcm.c sound/pci/pcxhr/pcxhr.c sound/pci/riptide/riptide.c sound/pci/rme32.c sound/pci/rme96.c sound/pci/trident/trident_main.c sound/pci/via82xx.c sound/pci/via82xx_modem.c sound/pci/ymfpci/ymfpci_main.c sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c sound/soc/at91/at91-pcm.c sound/soc/davinci/davinci-pcm.c sound/soc/omap/omap-pcm.c sound/soc/pxa/pxa2xx-pcm.c sound/soc/s3c24xx/s3c24xx-pcm.c sound/usb/usbaudio.c Please note that the dmix plugin does not support pausing even if the hardware device does. HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Audio hardware with pause support
Thanks for the hint, Clemens. If I interpret this information correctly, then it seems that many different sound drivers actually support pause, and consequently many audio architectures have the functionality as well (or it is emulated by the drivers). The real problem is the dmix plugin. In particular, I see hda_intel in the list. When I used ALSA with HDA Intel onboard hardware, pause was not available, probably due to the dmix plugin being used. So my next questions are: - What is the dmix plugin and what are the benefits of using it? - Is it possible to disable the dmix plugin? - What consequences does disabling the dmix plugin have? What essential features of ALSA will be missing without it? Best regards, Florian Clemens Ladisch wrote: Florian Winter wrote: Is there another way to determine whether a certain hardware supports snd_pcm_pause without having to test the hardware? $ grep -rl SNDRV_PCM_INFO_PAUSE sound sound/arm/pxa2xx-pcm.c sound/arm/sa11xx-uda1341.c sound/core/pcm_native.c sound/drivers/vx/vx_pcm.c sound/pci/als300.c sound/pci/atiixp.c sound/pci/au88x0/au88x0_pcm.c sound/pci/cmipci.c sound/pci/cs4281.c sound/pci/cs5535audio/cs5535audio_pcm.c sound/pci/echoaudio/darla20.c sound/pci/echoaudio/darla24.c sound/pci/echoaudio/echo3g.c sound/pci/echoaudio/gina20.c sound/pci/echoaudio/gina24.c sound/pci/echoaudio/indigo.c sound/pci/echoaudio/indigodj.c sound/pci/echoaudio/indigoio.c sound/pci/echoaudio/layla20.c sound/pci/echoaudio/layla24.c sound/pci/echoaudio/mia.c sound/pci/echoaudio/mona.c sound/pci/emu10k1/emupcm.c sound/pci/ens1370.c sound/pci/es1968.c sound/pci/fm801.c sound/pci/hda/hda_intel.c sound/pci/ice1712/ice1712.c sound/pci/ice1712/ice1724.c sound/pci/intel8x0.c sound/pci/intel8x0m.c sound/pci/maestro3.c sound/pci/mixart/mixart.c sound/pci/nm256/nm256.c sound/pci/oxygen/oxygen_pcm.c sound/pci/pcxhr/pcxhr.c sound/pci/riptide/riptide.c sound/pci/rme32.c sound/pci/rme96.c sound/pci/trident/trident_main.c sound/pci/via82xx.c sound/pci/via82xx_modem.c sound/pci/ymfpci/ymfpci_main.c sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c sound/soc/at91/at91-pcm.c sound/soc/davinci/davinci-pcm.c sound/soc/omap/omap-pcm.c sound/soc/pxa/pxa2xx-pcm.c sound/soc/s3c24xx/s3c24xx-pcm.c sound/usb/usbaudio.c Please note that the dmix plugin does not support pausing even if the hardware device does. HTH Clemens -- Florian Winter Software-Entwickler - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd-dummy 2nd card
Le Tue, 10 Jun 2008 17:47:58 +0300, alexander merkulov [EMAIL PROTECTED] a écrit : need to setup 2nd dummy card how to do it? You must add some device definitions in /etc/modules.d/alsa (or whatever file your distribution is using): ## ALSA portion alias snd-card-0 snd-... options snd-... index=0 alias snd-card-1 snd-dummy options snd-dummy index=1 ## OSS/Free portion alias sound-slot-0 snd-card-0 alias sound-slot-1 snd-card-1 # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss ## OSS/Free portion - card #2 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-12 snd-pcm-oss ## alias alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=2 Don't forget to run update-modules. I use this for virmidi instead of dummy, but I think that must work the same for the dummy sound card. Cheers, Dominique -- Dominique Michel Mes 3 projets préférés auxquels je contribue: * FVWM-Crystal, le bureau basé sur FVWM: http://fvwm-crystal.org * AlsaPlayer, le lecteur audio avec contrôle de vitesse en continu: www.alsaplayer.org * L'overlay pour la MAO sous gentoo: http://proaudio.tuxfamily.org/wiki/index.php?title=Main_Page - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output latency
Did you try the settings in /etc/security/limits.conf suggested on the Frinika front page? (http://frinika.sourceforge.net). I noticed quite some difference in delay for Terratec Aureon 5.1 Fun cards using JavaSound. Helge F. On Wed, Jun 11, 2008 at 10:11 AM, Clemens Ladisch [EMAIL PROTECTED] wrote: Alexander Carôt wrote: 3.) Rather than using a double buffer for the playout wouldn't it be possible to choose only one physical playout buffer and parse the captured data in right at the interrupt. It's unlikely that any code could be fast enough to write the entire buffer before the hardware starts reading from the buffer; in fact, the hardware is likely to read the first sample from the buffer at the same time it is triggering the interrupt for the previous period. To reduce output latency, just reduce the output buffer size. For example, you could use a buffer with 128 samples (with 2 64-sample periods). HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output latency
Hi Helge, what we actually discuss is more a principle of the driver related to buffer management and in how far this could reduce the latency. But thanks anyway ! -- A l e x Original-Nachricht Datum: Wed, 11 Jun 2008 18:06:27 +0200 Von: Helge Fredriksen [EMAIL PROTECTED] An: Clemens Ladisch [EMAIL PROTECTED] CC: alsa-user@lists.sourceforge.net, Alexander Carôt [EMAIL PROTECTED] Betreff: Re: [Alsa-user] Output latency Did you try the settings in /etc/security/limits.conf suggested on the Frinika front page? (http://frinika.sourceforge.net). I noticed quite some difference in delay for Terratec Aureon 5.1 Fun cards using JavaSound. Helge F. On Wed, Jun 11, 2008 at 10:11 AM, Clemens Ladisch [EMAIL PROTECTED] wrote: Alexander Carôt wrote: 3.) Rather than using a double buffer for the playout wouldn't it be possible to choose only one physical playout buffer and parse the captured data in right at the interrupt. It's unlikely that any code could be fast enough to write the entire buffer before the hardware starts reading from the buffer; in fact, the hardware is likely to read the first sample from the buffer at the same time it is triggering the interrupt for the previous period. To reduce output latency, just reduce the output buffer size. For example, you could use a buffer with 128 samples (with 2 64-sample periods). HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Dipl.-Ing. Alexander Carôt PhD Candidate Email : [EMAIL PROTECTED] Tel.: +49 (0)177 5719797 Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output latency
Hi Jochen, 2. use a lower frame size, than my codec/systems framing. (e.g. 128 instead of 256, but still transmit 256 in one pass) Yes - a good idea, however, sometimes depending on the actual machine and OS (or even low-latency patches) problems might occur when running below 256 samples/frame - especially when running the coding and network functionality. This is why I am trying to approach a solution that simply gets rid of the playout latency. You know that you can have one frame(-1 sample) additional delay, depending on the absolute framing of sender and receiver sound cards? E.g. the sender starts to grab samples at sample time 0, encodes it and sends it to the receiver. lets say the receiver has exactly the same framing as the sender and starts its play-out at sample time 0 as well. The transmission/decoding/etc needs one sample time and so the receiver has the current frame available at sample time 1. It just played out the last frame (at time 0) and the current frame needs to wait framesize-1 samples, and therefore almost one frame of additional delay. Regarding a double/multi buffers in ALSA, I have not played around yet, but maybe the threshold values for starting the play-out are a good starting point. Did you try those? You know, at play-out, the delay is not introduced by the number of 'periods', but the time the play-out starts. I once implemented a soundcard synchronisation which parses an incoming buffer slightly before the receiving card´s interrupt. The problem in that context is that the receiver is also a sender, which messes up this principle into the other direction. In turn this means that a sync at T/2 (half of the period) for both cards is most useful since it provides an equal delay for both ends. The reason why this practically still doen't work is that depending on the actual soundcard very often there is a slight soundcard drift which messes up this synchronisation over time anyway. It would be great to keep up the discussion. Thanks -- A l e x Best Regards, jochen On 10.Jun, 2008, at 17:02 , Alexander Carôt wrote: If I understand your question correctly, it is because they use two different buffers. If you aren't trying to play the capture buffer, it would wreak havoc to try to use it for playback while capture is going on. So there is a buffer for capture and a buffer for playback. And each has a latency. Someone who understands the system better might be able to give a better explanation, or even suggest a workaround. Sure - there is one input double buffer and one output double buffer. However, I wonder if there is a way to (somehow) get rid of the latter. The idea is the following : 1.) Of course there has to be an input double buffer which generates the desired block of samples. 2.) Once this is generated it takes 2,6 ms to generate the next one. 3.) Rather than using a double buffer for the playout wouldn't it be possible to choose only one physical playout buffer and parse the captured data in right at the interrupt. Or would this approach lead to timing and synchronisation problems ? I can see that either a parsing slightly too fast or too slow would result in wrong data but anyway - just an idea. What do you think ? Best regards -- A l e x -- Dipl.-Ing. Alexander Carôt PhD Candidate Email : [EMAIL PROTECTED] Tel.: +49 (0)177 5719797 Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user - jochen gpg: 1024D/FFE35929 -- Dipl.-Ing. Alexander Carôt PhD Candidate Email : [EMAIL PROTECTED] Tel.: +49 (0)177 5719797 GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED] - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Output sample rate
Where is my output sample rate defined? I'm trying to make sure mpd isn't resampling my music before it's sent to the USB DAC. - Grant - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
Grant wrote: Where is my output sample rate defined? I'm trying to make sure mpd isn't resampling my music before it's sent to the USB DAC. If you are playing audio that is in recognized format the rate is defined within the audio file itself and was set at time of creation. If you want to look at this you can use audacity to load the file and it will tell you the frame rate (lower left corner) or you can use sox with the stat command to get stats for a file (sox file dummyout.wav stat, man sox for more info). CD quality is 44100 as that is the standard. Most digital video sound is at 48000. Those are the two most common frame rates in use today. - Grant - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
-Original Message- From: Grant [EMAIL PROTECTED] To: alsa-user@lists.sourceforge.net Date: Wed, 11 Jun 2008 15:35:42 -0700 Subject: [Alsa-user] Output sample rate Where is my output sample rate defined? I'm trying to make sure mpd isn't resampling my music before it's sent to the USB DAC. - Grant I initiated a similar thread recently. The short answer - nowhere. As I was explained, an application MAY set sample rate. I think in practical terms use JACK since at launch you must specify sample rate and then it stays constant until you terminate JACK. Regards, Sergei. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
On 12-06-08 00:35, Grant wrote: Where is my output sample rate defined? I'm trying to make sure mpd isn't resampling my music before it's sent to the USB DAC. Check /proc/asound/card0/pcm0p/sub0/hw_params while mpd is playing (for a suitable value of (0,0,0) ofcourse). Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
On 12-06-08 02:17, Sergei Steshenko wrote: From: Grant [EMAIL PROTECTED] Where is my output sample rate defined? I'm trying to make sure mpd isn't resampling my music before it's sent to the USB DAC. I initiated a similar thread recently. The short answer - nowhere. As I was explained, an application MAY set sample rate. Well, no, an ALSA application _must_ set sample rate -- to the rate of the data it then starts feeding the device. The thing is and was just that without resampling anything, it's the data and not the card that determines this rate. The sampling rate is a property inherent to the data. Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
-Original Message- From: Rene Herman [EMAIL PROTECTED] To: Sergei Steshenko [EMAIL PROTECTED] Date: Thu, 12 Jun 2008 04:12:58 +0200 Subject: Re: [Alsa-user] Output sample rate The sampling rate is a property inherent to the data. Rene. ??? Are trying to tell me that sample rate is inherent to analog source connected to microphone or line input ? Regards, Sergei. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd-dummy 2nd card
On 11-06-08 17:41, Dominique Michel wrote:
Le Tue, 10 Jun 2008 17:47:58 +0300,
alexander merkulov [EMAIL PROTECTED] a écrit :
need to setup 2nd dummy card
how to do it?
You must add some device definitions in /etc/modules.d/alsa (or whatever file
your distribution is using):
## ALSA portion
alias snd-card-0 snd-...
options snd-... index=0
alias snd-card-1 snd-dummy
options snd-dummy index=1
options snd-dummy index=1,2 enable=1,1 was more the issue, but:
## OSS/Free portion
alias sound-slot-0 snd-card-0
alias sound-slot-1 snd-card-1
--- 8 --- cut here --- 8 ---
# OSS/Free portion - card #1
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
## OSS/Free portion - card #2
alias sound-service-1-0 snd-mixer-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-12 snd-pcm-oss
please note that these aliases have been not necessary for some time now
as the modules export them themselves:
# modinfo snd-{pcm,mixer,seq}-oss | grep alias
alias: sound-service-?-12
alias: sound-service-?-3
alias: sound-service-?-0
alias: sound-service-?-8
alias: sound-service-?-1
## alias
alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss
and these should be automatic through the soundcore module handling
char-major-14 and it requesting sound-slot-? and sound-service-?-?.
So, you can loose everything but the snd-card-?/sound-slot-? aliases.
# Set this to the correct number of cards.
options snd cards_limit=2
Or not. I once wasted time tracking down a non-bug due to forgetting I
had limited the number of cards like this. The default is 8...
Rene.
-
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Re: [Alsa-user] Output sample rate
On 12-06-08 05:52, Sergei Steshenko wrote: From: Rene Herman [EMAIL PROTECTED] The sampling rate is a property inherent to the data. Are trying to tell me that sample rate is inherent to analog source connected to microphone or line input ? Oh, not again... please get a clue. DATA. Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
-Original Message- From: Rene Herman [EMAIL PROTECTED] To: Sergei Steshenko [EMAIL PROTECTED] Date: Thu, 12 Jun 2008 05:55:40 +0200 Subject: Re: [Alsa-user] Output sample rate On 12-06-08 05:52, Sergei Steshenko wrote: From: Rene Herman [EMAIL PROTECTED] The sampling rate is a property inherent to the data. Are trying to tell me that sample rate is inherent to analog source connected to microphone or line input ? Oh, not again... please get a clue. DATA. Rene. Are you trying to tell me that all soundcards support different (at the same time) capture and playback sample rates ? Are you trying to tell me that in case soundcard is used to play back analog data from a TV or radio card it takes sample rate from the air ? Or from input analog signal ? Yes again - to me ALSA's sample rate implementation looks quite illogical - IMO it should be the other way round - user first mandates sample rate, and then playback sources adapt through resampling if necessary. Regards, Sergei. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
On 12-06-08 06:30, Sergei Steshenko wrote: Yes again - to me ALSA's sample rate implementation looks quite illogical - IMO it should be the other way round - user first mandates sample rate, and then playback sources adapt through resampling if necessary. Great setup once we have infinitely fast computers to do the resampling at any quality level we feel like in realtime. Meanwhile over here in the real world we'll continue on our imperfect attempts to make things work though. Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
Its very simple. Most sound devices support a number of sample rates. Common ones include 16 bit @ 44.1khz, 16-bit @ 48khz, 24-bit @ 96 khz etc. Only one application has exclusive control over the sound hardware at any time. Whatever rate that application opens the soundcard at, is the rate that is used to send data to the card. If the application with exclusive control over the sound hardware is a mixer-type daemon (e.g. dmix, esd, artsd, etc.) then all audio streams are converted by this application and sent to the sound card using the sample rate it opened the hardware with. Does that make sense? -Pete On 12-06-08 06:30, Sergei Steshenko wrote: Yes again - to me ALSA's sample rate implementation looks quite illogical - IMO it should be the other way round - user first mandates sample rate, and then playback sources adapt through resampling if necessary. Great setup once we have infinitely fast computers to do the resampling at any quality level we feel like in realtime. Meanwhile over here in the real world we'll continue on our imperfect attempts to make things work though. Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
On 12-06-08 06:53, Pete Black wrote: Its very simple. Most sound devices support a number of sample rates. Common ones include 16 bit @ 44.1khz, 16-bit @ 48khz, 24-bit @ 96 khz etc. Only one application has exclusive control over the sound hardware at any time. Whatever rate that application opens the soundcard at, is the rate that is used to send data to the card. If the application with exclusive control over the sound hardware is a mixer-type daemon (e.g. dmix, esd, artsd, etc.) then all audio streams are converted by this application and sent to the sound card using the sample rate it opened the hardware with. Does that make sense? dmix isn't an application (it just mixes behind the scenes) but basically, yes, that is how things work. And given that dmix is default these days (if the card doesn't support hardware mixing that is) Sergei's notion of how things might work is how things _do_ work in practice right now for anyone who doesn't care enough about his audio to disable dmix. The first A in ALSA stands for advanced though and you can bet your testicles that many advanced users do not look favourably upon resampling and dmix... Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
Computer audio and sample rate issues are popping up everywhere, driven by the desire for high quality audio on PC's finally. On Windows and Mac's its even harder to get it right. In Alsa (and PC audio architecture in general) the system has a default sample rate, usually set by the driver it seems, and usually 48 KHz. Normally all the audio is resampled to this rate to permit mixing audio from different sources (the system alert noises with your MP3 playback for instance). However for those of us trying to get very high quality audio out of the PC it's possible to get the driver, the audio player and the card to cooperate and play the stream at its original bit perfect' sample rate. It requires alignment of the audio player app (Xine in my case), Alsa, the driver for the card (the latest version of the Juli@ driver) and it all seems to work right after many hours of tweaking. I have NOT played with USB audio. For 44.1 and 48 it should be pretty straightforward, the higher sample rates will require chip/card specific drivers something never easy on Linux. On a Mac you pretty much go deep into the system and set the sample rate for the whole system and change it for every change in source rate. On a PC you don't want Vista (yet. . . ) and you need to use ASIO and a lot of fiddling. Which is why I'm using Linux. Demian Martin Product Design Services - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
dmix *is* an application, conceptually it is not different to esd, artsd or pulseaudio opening the ALSA hardware directly. It just happens to be an ALSA plugin and is a part of the signal chain that ALSA-lib sets up by default. You can either: a) configure dmix using a custom .asoundrc to use the rate you want and restart ALSA. b) configure an application that will open the sound card with the rate you want, and set it to open the sound device directly e.g. hw:0,0 in ALSA terms. The sample rate that is used is determined, plain and simple, by whichever application has exclusive control over the sound card. dmix isn't an application (it just mixes behind the scenes) but basically, yes, that is how things work. And given that dmix is default these days (if the card doesn't support hardware mixing that is) Sergei's notion of how things might work is how things _do_ work in practice right now for anyone who doesn't care enough about his audio to disable dmix. The first A in ALSA stands for advanced though and you can bet your testicles that many advanced users do not look favourably upon resampling and dmix... Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
On 12-06-08 07:13, Demian Martin wrote: Computer audio and sample rate issues are popping up everywhere, driven by the desire for high quality audio on PC's finally. On Windows and Mac's its even harder to get it right. In Alsa (and PC audio architecture in general) the system has a default sample rate, usually set by the driver it seems, and usually 48 KHz. Well, no. I have to comment on this since it gives the wrong impression about the design of things. ALSA drivers do not have or set a default sampling rate and whether or not the system does depends on you considering the dmix plugin intrinsic enough to be the system. It is enabled by default (as also said in another message just now, if the card doesn't support mixing in hardware) which means you might be tempted to view it this way, but dmix is still only a fairly recent addition and implemented just under prressure of desktop sound needing to Just Work (after which everybody goes off and uses soundserver apps exclusively anyway it seems, but oh well...). dmix is a plugin which indeed resamples to 48 kHz. It's easily disabled though by just using the hw ALSA device instead of the ALSA device named default. To disable dmix from that device is also just a few config file edits away... Normally all the audio is resampled to this rate to permit mixing audio from different sources (the system alert noises with your MP3 playback for instance). However for those of us trying to get very high quality audio out of the PC it's possible to get the driver, the audio player and the card to cooperate and play the stream at its original bit perfect' sample rate. It requires alignment of the audio player app (Xine in my case), Alsa, the driver for the card (the latest version of the Juli@ driver) and it all seems to work right after many hours of tweaking. You could've just disabled dmix... :-) (by editing a card config file under /usr/share/alsa/cards or overriding things in /etc/alsa.conf or .asoundrc). Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output sample rate
On 12-06-08 07:17, Pete Black wrote: dmix *is* an application, conceptually it is not different [ ... ] Let's call it a conceptlication then. As said, your basic description was correct. Rene. - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
