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author | Marton Balint <cus@passwd.hu> | 2011-08-17 01:28:23 +0200 |
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committer | Marton Balint <cus@passwd.hu> | 2011-10-02 22:23:11 +0200 |
commit | 1dd3c473a2096e60b2e5a765eaabb378c34b3537 (patch) | |
tree | 9f682bdd4a774a375a980d552ebfb521a66a0e3c /libavcodec/fmtconvert.h | |
parent | 60aaf044f3fdbeb1cf702b144cb0597ce4988020 (diff) | |
download | ffmpeg-streaming-1dd3c473a2096e60b2e5a765eaabb378c34b3537.zip ffmpeg-streaming-1dd3c473a2096e60b2e5a765eaabb378c34b3537.tar.gz |
ffplay: use libswresample instead of av_audio_convert
Previously ffplay expected SDL_AudioOpen to provide the requested sample rate
and channel number. This is no longer a requirement because this patch replaces
the audio convert function with libswresample's swr_convert which is capable of
handling different sample formats, sample rates and different number of
channels and different channel layouts.
The patch also removes the hardcoded 16bit samples assumption and uses
av_get_bytes_per_sample almost everywhere. The only exceptions are
the update_sample_display and video_audio_display functions, it
seemed too much of a headache to make them generic.
We also fix a tiny bug in sdl_audio_callback, we ensure that the number of
bytes when we put silence in the buffer is a multiple of the frame size.
Diffstat (limited to 'libavcodec/fmtconvert.h')
0 files changed, 0 insertions, 0 deletions