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

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[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?


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