Re: FIle plugin and format change

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On Sat, 22 Dec 2007, Carlos Hernandez wrote:

> well... I really want to use another application(mp4live)... not arecord.
>
> So this application just ask me for : "sample rate", "channels" and
> what Alsa device to use.

Except that mp4live does not use alsa, it uses oss if I read the data right

List of Media Sources

V4L	video_v4l_source.cpp
Acquires YUV12 images from a V4L (Video For Linux) device
V4L2	video_v4l2_source.cpp
Acquires YUV12 images from a V4L2 (Video For Linux) device (recommended)
OSS	audio_oss_source.cpp
Acquires PCM16 audio samples from an OSS (Open Sound System) device
self	loop_source.cpp
Acquires raw YUV or PCM from main process





>
> I've tell it "plugfile" device, 44100 as sample rate, and stereo(2
> channels). As you can see I cannot specify the format.
>
> When I run mp4live it says:
>
> mp4live: pcm_params.c:2351: sndrv_pcm_hw_params: Assertion `err >= 0' failed.
>
> I dont know what to do.
>
>
> Thanks
>
>
>
>
> On 22/12/2007, Bill Unruh <unruh@xxxxxxxxxxxxxx> wrote:
>> On Sat, 22 Dec 2007, Carlos Hernandez wrote:
>>
>>> I know... and that's the reason why I can specify that, using the asound.conf.
>>>
>>> The source audio file is in 16 bit, 44100, stereo...  but when I
>>> record with arecord, it does with 8bit, 8k, mono.
>>>
>>> So I must change this before arecord "record" the audio... where? .. I
>>> suppose this should be in asound.conf (or .asoundrc)... setting in the
>>> plug parameters "format", "channels" and "rate"... this way :
>>
>> arecord -f cd >outfile
>>
>>>
>>> pcm.plugfile{
>>>       type plug
>>>       slave {
>>>               pcm infile
>>>               format S16_LE
>>>               channels 2
>>>               rate 44100
>>>       }
>>> }
>>>
>>> but this does not work...
>>>
>>>
>>> thanks anyway
>>>
>>>
>>>
>>> On 22/12/2007, Lee Revell <rlrevell@xxxxxxxxxxx> wrote:
>>>> On Dec 21, 2007 9:39 PM, Carlos Hernandez <dspcarlos@xxxxxxxxx> wrote:
>>>>> Here the "format", "rate" and "channels" are set up, and these are the
>>>>> same as the original audio file.  Then I suppose I would not have to
>>>>> specify these parameters with arecord.
>>>>> running arecord:
>>>>>
>>>>> carlos@chaos:~$ arecord -D plugfile out.wav
>>>>> Recording WAVE 'out.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
>>>>>
>>>>
>>>> arecord does not do any kind of format detection on its input.  It
>>>> records with whatever format you tell it to (in this case, the default
>>>> of 8Khz 8-bit mono)
>>>>
>>>> Lee
>>>>
>>>
>>>
>>>
>>
>> --
>> William G. Unruh   |  Canadian Institute for|     Tel: +1(604)822-3273
>> Physics&Astronomy  |     Advanced Research  |     Fax: +1(604)822-5324
>> UBC, Vancouver,BC  |   Program in Cosmology |     unruh@xxxxxxxxxxxxxx
>> Canada V6T 1Z1     |      and Gravity       |  www.theory.physics.ubc.ca/
>>
>
>
>

-- 
William G. Unruh   |  Canadian Institute for|     Tel: +1(604)822-3273
Physics&Astronomy  |     Advanced Research  |     Fax: +1(604)822-5324
UBC, Vancouver,BC  |   Program in Cosmology |     unruh@xxxxxxxxxxxxxx
Canada V6T 1Z1     |      and Gravity       |  www.theory.physics.ubc.ca/

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