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authorStefano Sabatini <stefano.sabatini-lala@poste.it>2011-02-14 10:59:14 +0100
committerJanne Grunau <janne-ffmpeg@jannau.net>2011-02-16 10:33:54 +0100
commit54955283655f918db4e9c8ec5a254f06a10646a6 (patch)
tree019873712e5a6d6c94c108792631f026d47b8b49 /libavcodec/avcodec.h
parent5d820db2f9c396373f2293e6b41944093a4766d5 (diff)
Apply minor cosmetics fixes to the av_audio_resample_init() doxy.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
Diffstat (limited to 'libavcodec/avcodec.h')
-rw-r--r--libavcodec/avcodec.h7
1 files changed, 4 insertions, 3 deletions
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 84c3b6fc79..3cc212f7cd 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3222,8 +3222,9 @@ typedef struct ReSampleContext ReSampleContext;
attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate);
#endif
+
/**
- * Initialize audio resampling context
+ * Initialize audio resampling context.
*
* @param output_channels number of output channels
* @param input_channels number of input channels
@@ -3231,9 +3232,9 @@ attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, i
* @param input_rate input sample rate
* @param sample_fmt_out requested output sample format
* @param sample_fmt_in input sample format
- * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
+ * @param filter_length length of each FIR filter in the filterbank relative to the cutoff frequency
* @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
+ * @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
* @return allocated ReSampleContext, NULL if error occured