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| author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-11-01 00:44:11 -0400 |
|---|---|---|
| committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-11-27 16:49:18 -0500 |
| commit | 7f534d11ed37dd03cbaa73eefbfa75696ff04f58 (patch) | |
| tree | 2c9e6e76122fe33798fe31ad03161f6a8a2d1e4b /libavresample/audio_convert.c | |
| parent | 28e1cf19aa3c2467ec02c670af05af2610deef8c (diff) | |
| download | ffmpeg-7f534d11ed37dd03cbaa73eefbfa75696ff04f58.tar.gz | |
lavr: do not pass sample count as a parameter to ff_audio_convert()
It will always be the number of samples in the input buffer, so just use that
directly instead of passing it as a separate parameter.
Diffstat (limited to 'libavresample/audio_convert.c')
| -rw-r--r-- | libavresample/audio_convert.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c index e9835c8e8b..dcf8a39b06 100644 --- a/libavresample/audio_convert.c +++ b/libavresample/audio_convert.c @@ -284,9 +284,10 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, return ac; } -int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len) +int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic = 1; + int len = in->nb_samples; /* determine whether to use the optimized function based on pointer and samples alignment in both the input and output */ |