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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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
On Wed, Jul 15, 2015 at 12:50 PM, Ran Shalit ransha...@gmail.com 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 executable
I
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 words,
On Fri, Oct 9, 2015 at 11:52 AM, Clemens Ladisch
<cladi...@googlemail.com> 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
On Thu, Oct 8, 2015 at 7:55 PM, chris hermansen <clherman...@gmail.com> wrote:
> Ran and list,
>
> On Oct 8, 2015 09:29, "Ran Shalit" <ransha...@gmail.com> wrote:
>>
>> Hello,
>>
>> I don't understand why rate is required when we play PCM fil
On Fri, Sep 11, 2015 at 9:42 AM, Clemens Ladisch
<cladi...@googlemail.com> 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
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
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
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
On Sat, Sep 19, 2015 at 11:20 AM, Ran Shalit <ransha...@gmail.com> 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 t
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
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