On Sun, Jan 19, 2020 at 4:28 PM Carl Eugen Hoyos <[email protected]> wrote:
> Am Mo., 20. Jan. 2020 um 01:22 Uhr schrieb Suhail Doshi < > [email protected]>: > > > Sure, do you know why ffmpeg cli seems to be able to encode interleaved > > raw audio but the C API only allows FLTP then? > > It (automatically) inserts the aresample filter into the filter chain. > > Please find out what top-posting means and avoid it here, Carl Eugen > Got it. So, I tried to resample my FLT audio into FLTP audio as well. I got a bit stuck. Here's my code: https://gist.github.com/Suhail/151e41f3eb226504c7cbd3b46c15729c (I didn't want to paste it here since it's long where I referenced this code heavily <https://github.com/FFmpeg/FFmpeg/blob/master/libavresample/avresample.h#L41> . What I do, as a test, is I read an entire PCM raw audio file into a buffer and then send that to the encoder. While the encoder doesn't output an error, it doesn't seem to output any valid AAC encoded audio either. Even after a flush, it seems to provide a substantially small amount of information that's invalid to play. I also tried sending it packets of raw audio captured from PulseAudio but received similar results. Any ideas? I feel like I am missing something fundamental. > _______________________________________________ > Libav-user mailing list > [email protected] > https://ffmpeg.org/mailman/listinfo/libav-user > > To unsubscribe, visit link above, or email > [email protected] with subject "unsubscribe".
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