On Fri, Oct 9, 2015 at 11:52 AM, Clemens Ladisch
wrote:
> Ran Shalit wrote:
>> I don't understand why rate is required when we play PCM file.
>
> The rate is _not_ required when we play PCM file.
>
> If the file header has the rate, aplay ignores the rate parameter.
&g
On Thu, Oct 8, 2015 at 7:55 PM, chris hermansen wrote:
> Ran and list,
>
> On Oct 8, 2015 09:29, "Ran Shalit" wrote:
>>
>> Hello,
>>
>> I don't understand why rate is required when we play PCM file.
>> The sample rate is already encoded in
Hello,
I don't understand why rate is required when we play PCM file.
The sample rate is already encoded in file header, so why would aplay
need this parameter ?
Thanks,
Ran
--
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On Sat, Sep 19, 2015 at 11:20 AM, Ran Shalit wrote:
> Hello,
>
> I'm using arecord as following:
> arecord --device=plughw:0,0 --duration=2 test.wav
> Anit is prints that it is using rate: 8000Hz
> But on opening it with audacity, or vlc, it shows that sample rate is
Hello,
I'm using arecord as following:
arecord --device=plughw:0,0 --duration=2 test.wav
Anit is prints that it is using rate: 8000Hz
But on opening it with audacity, or vlc, it shows that sample rate is 44100.
Is there any explaination for this behaviour ?
Regards,
Ran
Hello,
I use arecord to record as done in the following line:
arecord --device=plughw:0,1 --duration=8 /mnt/nfs/test.wav
But on playing the recorded file (vlc or audacity) it is played faster
then the real speed:
https://drive.google.com/file/d/0B22GsWueReZTUnFvYXZRNjBiZXM/view?usp=sharing
What
Hello,
I am using 2 cards for sound.
I have configured the crads/hw in kernel for the custom board.
With card0 , arecord seems OK (not yet checked the result record wav
file), but with card1 it hangs in (strace output):
poll([{fd=4, events=POLLIN|POLLERR|POLLNVAL}], 1, -1
Does anyone have any idea
On Fri, Sep 11, 2015 at 9:42 AM, Clemens Ladisch
wrote:
> Ran Shalit wrote:
>> asoc: failed to add dapm kcontrol Right HPCOM Mixer Line2L Bypass Switch: -16
>> asoc: failed to add kcontrol Right Line2R Mux
>> ...
>> What is the reason for the "asoc: fail
Hello,
I'm using 2 codecs (tlv320aic3x-codec) each which both are registered
into same sound card, each as a single device.
After boot I have the following devices:
root@dm814x:~# cat /proc/asound/devices
0: [ 0] : control
16: [ 0- 0]: digital audio playback
17: [ 0- 1]: digital audio play
On Wed, Jul 15, 2015 at 12:50 PM, Ran Shalit wrote:
> Hello,
>
> I cross-compile alsa-project for arm target.
> It seems to work find.
> I then wanted to compile test examples:
> make pcm
> CC pcm.o
> CCLD pcm
> But It seems to create just object .o, not
>
>
>
> "Dual mono" is indeed an AAC-related concept.
>
> When you record multiple channels using ALSA, they are always independent,
> i.e., ALSA does not make any assumptions about the sample values in the
> channels (or, for that matter, about consecutive samples in the same
> channel). In other
Hello,
I cross-compile alsa-project for arm target.
It seems to work find.
I then wanted to compile test examples:
make pcm
CC pcm.o
CCLD pcm
But It seems to create just object .o, not executable
I see that there is additional pcm script file in the test folder.
Does anyone know
Hello,
I am new with ALSA project.
I would like to ask if dual mono, is used in ALSA APIs, or wether it is the
same as stereo.
Thank you,
Ran
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