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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:10:38 +0200 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:40:12 +0200 |
| commit | f8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch) | |
| tree | 0ebda51a6ba23d790da30a7168870928954da395 /libavresample/internal.h | |
| parent | bf5386385dc504a076453ad58f61f808677be747 (diff) | |
| parent | 5467742232c312b7d61dca7ac57447f728d8d6c9 (diff) | |
| download | ffmpeg-f8911b987de4a84ff8ae92f41ff492ece4acadb9.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
mss3: use standard zigzag table
mss3: split DSP functions that are used in MTS2(MSS4) into separate file
motion-test: do not use getopt()
tcp: add initial timeout limit for incoming connections
configure: Change the rdtsc check to a linker check
avconv: propagate fatal errors from lavfi.
lavfi: add error handling to filter_samples().
fate-run: make avconv() properly deal with multiple inputs.
asplit: don't leak the input buffer.
af_resample: fix request_frame() behavior.
af_asyncts: fix request_frame() behavior.
libx264: support aspect ratio switching
matroskadec: honor error_recognition when encountering unknown elements.
lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
lavr: resampling: add filter type and Kaiser window beta to AVOptions
lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
lavr: mix: validate internal sample format in ff_audio_mix_init()
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/libx264.c
libavfilter/audio.c
libavfilter/split.c
libavformat/tcp.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavresample/internal.h')
| -rw-r--r-- | libavresample/internal.h | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/libavresample/internal.h b/libavresample/internal.h index fa9499a8ef..7b7648f0be 100644 --- a/libavresample/internal.h +++ b/libavresample/internal.h @@ -50,6 +50,8 @@ struct AVAudioResampleContext { int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */ int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */ double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */ + enum AVResampleFilterType filter_type; /**< resampling filter type */ + int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ int in_channels; /**< number of input channels */ int out_channels; /**< number of output channels */ |