[Libav-user] avcodec_decode_audio4() help
ggarra13 at gmail.com
Tue May 7 00:47:08 CEST 2013
I placed a print statement with:
> const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
> and the return value for 1.0.6 is s16, while for 1.1.4 is fltp, which
> I assume is float planar.
> Does this mean I need to use the swresample library to resample the
> sound to s16 which the audio card can play?
When I run the video under ffmpeg I got:
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo,
s16, 159 kb/s
Stereo, s16. Why am I getting float data in the newer ffmpegs?
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