[Libav-user] Question about: audio_codec, bit_rate, sample_rate, channels, channel_layouts

[Denis Gottardello]

Hi, my *source* stream is
Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
and in order to store it in an avi file I have created a similar *destination* audio stream 
like that:
Stream #0:1: Audio: pcm_alaw, 8000 Hz, 1 channels, s16, 64 kb/s
I can hear the audio track by a pcm decoder and the audio is ok.
In order to hear the source stream I have to reproduce a buffer that it is length *320* 
byte.
After having put a printf function in ffmpeg library implementation I see that
nb_samples= 10000
linesize[0]= 20032
Why nb_samples is not 320 / 2 but it is 10000?


+39.347.4070897
http://www.labcsp.com[1]
http://www.denisgottardello.it[2]
GMT+1
Skype: mrdebug

--------
[1] http://www.labcsp.com
[2] http://www.denisgottardello.it
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://ffmpeg.org/pipermail/libav-user/attachments/20200510/83245211/attachment.html>