[Libav-user] Reducing sample rate and channels for encoding

[Goo Creations]

Hi,

I'm relatively new to FFMpeg. I have a buffer in memory of decoded samples
(44100Hz, 2 channels) and I want to encode them to 22050Hz mono. Hence, the
size of the buffer should be 4 times smaller.

If I pass my samples to av_encode_audio2, should I accommodate for this
reduction or will FFMpeg handle this? For example if I want to encode from
stereo to mono, should I remove every second sample before passing the
buffer to av_encode_audio2, or will this function automatically remove
every second sample for me?

Thanks
Christoph
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