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authorMichael Niedermayer <michaelni@gmx.at>2009-02-09 15:49:06 +0000
committerMichael Niedermayer <michaelni@gmx.at>2009-02-09 15:49:06 +0000
commit990058bdc7a63f7a488179367b859307e99adf34 (patch)
tree0e14337a31e474a90fae26ca172f79a13668a1a3
parent3136fcf45ec7c8fc45171f628ecf5001f4d486a7 (diff)
downloadffmpeg-990058bdc7a63f7a488179367b859307e99adf34.tar.gz
Try to doxyfy av_resample_init(). I hope I did not misremember anything as I
just briefly checked the source. Originally committed as revision 17100 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/avcodec.h5
1 files changed, 5 insertions, 0 deletions
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 36a40861af..edc3adeef1 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -2452,6 +2452,11 @@ void audio_resample_close(ReSampleContext *s);
/**
* Initializes an audio resampler.
* Note, if either rate is not an integer then simply scale both rates up so they are.
+ * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
+ * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
+ * @param linear If 1 then the used FIR filter will be linearly interpolated
+ between the 2 closest, if 0 the closest will be used
+ * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
*/
struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);