[alsa-devel] Arecord issue
Hi, I am working on a 2.6.37 kernel and on a DM365 davinci board with a tlv320aic3254 audio codec (i2c control lines, i2s data transfer) I wanted to get the audio up on my board so got the alsa-lib and alsa-utils 1.0.25 compiled them and got the aplay to work fine. When I got to working on the arecord things haven't worked fine. I have traced all the calls and have reached so far; codec gets set properly davinci_pcm_trigger executes with success snd_pcm_sw_params() executes successfully. aplay calls begin_wave() aplay calls pcm_read() that invokes snd_pcm_readi() Finally my app seems to not go beyond the snd_pcm_start()
Any assistance shall be highly appreciated!
Thanks in advance, N.Kartik
On 11.10.2012 14:54, kartik natarajan wrote:
I am working on a 2.6.37 kernel and on a DM365 davinci board with a
tlv320aic3254 audio codec (i2c control lines, i2s data transfer) I wanted to get the audio up on my board so got the alsa-lib and alsa-utils 1.0.25 compiled them and got the aplay to work fine. When I got to working on the arecord things haven't worked fine. I have traced all the calls and have reached so far; codec gets set properly davinci_pcm_trigger executes with success snd_pcm_sw_params() executes successfully. aplay calls begin_wave() aplay calls pcm_read() that invokes snd_pcm_readi() Finally my app seems to not go beyond the snd_pcm_start()
I've been working on a custom AM33xx boards lately which features the same digital audio interface than the Davincis do.
I've been running into similar issues when I tested the recording channels, and hooking up a logic analyzer showed that the clocks were not driven by the CPU. That causes the DMA engine not receive any samples (as the frame and bit clocks are missing), which will eventually lead to a timeout.
This issue was worked around for now by starting the playback first and then the record. Unfortunately though, a simply "arecord | aplay" does not suffice, as aplay won't start before it received any samples from its input side, so you have a deadlock here.
So I wrote a very simple test tool in order to prove that the overall system is working and published it here:
https://github.com/zonque/simple-alsa-loop
For the application of this board I'm working on, there's no case where we would record anything but not use the output, so this workaround will do for me. If you have other requirements, you might need to dig deeper. Note that the driver has seem quite some refactorization in 3.7.
Daniel
Thanks! At about the same time I check the behaviour on the oscilloscope and inferred the same that there is no clock on the bus when arecord is executed. A little digging further, I see that the codec chip is set to master in both cases, aplay and arecord. But somehow it drives the clock in the former case and not in the other. So this problem is not related to either of aplay/ arecord but more of the PCM level behaviour, as it is at this level the master for the bus is decided. I'll get back with any information I get on this. Your mail has just made my findings easier. Thanks again.
On Thu, Oct 11, 2012 at 6:36 PM, Daniel Mack zonque@gmail.com wrote:
On 11.10.2012 14:54, kartik natarajan wrote:
I am working on a 2.6.37 kernel and on a DM365 davinci board with a
tlv320aic3254 audio codec (i2c control lines, i2s data transfer) I wanted to get the audio up on my board so got the alsa-lib and
alsa-utils
1.0.25 compiled them and got the aplay to work fine. When I got to working on the arecord things haven't worked fine. I have traced all the calls and have reached so far; codec gets set properly davinci_pcm_trigger executes with success snd_pcm_sw_params() executes successfully. aplay calls begin_wave() aplay calls pcm_read() that invokes snd_pcm_readi() Finally my app seems to not go beyond the snd_pcm_start()
I've been working on a custom AM33xx boards lately which features the same digital audio interface than the Davincis do.
I've been running into similar issues when I tested the recording channels, and hooking up a logic analyzer showed that the clocks were not driven by the CPU. That causes the DMA engine not receive any samples (as the frame and bit clocks are missing), which will eventually lead to a timeout.
This issue was worked around for now by starting the playback first and then the record. Unfortunately though, a simply "arecord | aplay" does not suffice, as aplay won't start before it received any samples from its input side, so you have a deadlock here.
So I wrote a very simple test tool in order to prove that the overall system is working and published it here:
https://github.com/zonque/simple-alsa-loop
For the application of this board I'm working on, there's no case where we would record anything but not use the output, so this workaround will do for me. If you have other requirements, you might need to dig deeper. Note that the driver has seem quite some refactorization in 3.7.
Daniel
Please do not top-post.
On 11.10.2012 16:44, kartik natarajan wrote:
Thanks! At about the same time I check the behaviour on the oscilloscope and inferred the same that there is no clock on the bus when arecord is executed. A little digging further, I see that the codec chip is set to master in both cases, aplay and arecord. But somehow it drives the clock in the former case and not in the other.
So this problem is not related to either of aplay/ arecord but more of the PCM level behaviour, as it is at this level the master for the bus is decided.
Please have a look at the latest changes to this driver found here:
https://git.kernel.org/?p=linux/kernel/git/broonie/sound.git;a=shortlog;h=re...
I changed many of these details to make the driver fit into more environments. If you find a solution to this problem, please post a patch that applies on top of this tree.
I'll get back with any information I get on this. Your mail has just made my findings easier. Thanks again.
You're welcome :)
Daniel
On Thu, Oct 11, 2012 at 6:36 PM, Daniel Mack <zonque@gmail.com mailto:zonque@gmail.com> wrote:
On 11.10.2012 14:54, kartik natarajan wrote: > I am working on a 2.6.37 kernel and on a DM365 davinci board with a > tlv320aic3254 audio codec (i2c control lines, i2s data transfer) > I wanted to get the audio up on my board so got the alsa-lib and alsa-utils > 1.0.25 compiled them and got the aplay to work fine. > When I got to working on the arecord things haven't worked fine. I have > traced all the calls and have reached so far; > codec gets set properly > davinci_pcm_trigger executes with success > snd_pcm_sw_params() executes successfully. > aplay calls begin_wave() > aplay calls pcm_read() that invokes snd_pcm_readi() > Finally my app seems to not go beyond the snd_pcm_start() I've been working on a custom AM33xx boards lately which features the same digital audio interface than the Davincis do. I've been running into similar issues when I tested the recording channels, and hooking up a logic analyzer showed that the clocks were not driven by the CPU. That causes the DMA engine not receive any samples (as the frame and bit clocks are missing), which will eventually lead to a timeout. This issue was worked around for now by starting the playback first and then the record. Unfortunately though, a simply "arecord | aplay" does not suffice, as aplay won't start before it received any samples from its input side, so you have a deadlock here. So I wrote a very simple test tool in order to prove that the overall system is working and published it here: https://github.com/zonque/simple-alsa-loop For the application of this board I'm working on, there's no case where we would record anything but not use the output, so this workaround will do for me. If you have other requirements, you might need to dig deeper. Note that the driver has seem quite some refactorization in 3.7. Daniel
-- Believe in your dreams, they have a strange way of coming true!!!
participants (2)
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Daniel Mack
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kartik natarajan