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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