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authorJustin Ruggles <justin.ruggles@gmail.com>2012-11-01 00:44:11 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2012-11-27 16:49:18 -0500
commit7f534d11ed37dd03cbaa73eefbfa75696ff04f58 (patch)
tree2c9e6e76122fe33798fe31ad03161f6a8a2d1e4b /libavresample/audio_convert.h
parent28e1cf19aa3c2467ec02c670af05af2610deef8c (diff)
downloadffmpeg-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.h')
-rw-r--r--libavresample/audio_convert.h7
1 files changed, 5 insertions, 2 deletions
diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h
index 2b8bface7d..bc27223140 100644
--- a/libavresample/audio_convert.h
+++ b/libavresample/audio_convert.h
@@ -72,13 +72,16 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
* examined to determine whether to use the generic or optimized conversion
* function (when available).
*
+ * The number of samples to convert is determined by in->nb_samples. The output
+ * buffer must be large enough to handle this many samples. out->nb_samples is
+ * set by this function before a successful return.
+ *
* @param ac AudioConvert context
* @param out output audio data
* @param in input audio data
- * @param len number of samples to convert
* @return 0 on success, negative AVERROR code on failure
*/
-int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len);
+int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
/* arch-specific initialization functions */