diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2015-02-03 01:27:26 +0300 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2015-02-03 01:27:26 +0300 |
commit | 9d7ae72725e16bc4b53ed6ccedf86d0ae2853809 (patch) | |
tree | de4e6c005d1913d15bf97f0424a87632b41e7aef | |
parent | c348a42dcc93c21cdc114769a29eba955bdb74f9 (diff) |
swresample: Use int instead of enum for fields which are accessed through AVOptions as int
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
-rw-r--r-- | libswresample/swresample_internal.h | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h index 6250921b8c..bb51272d6f 100644 --- a/libswresample/swresample_internal.h +++ b/libswresample/swresample_internal.h @@ -53,7 +53,7 @@ typedef struct AudioData{ } AudioData; struct DitherContext { - enum SwrDitherType method; + int method; int noise_pos; float scale; float noise_scale; ///< Noise scale @@ -106,10 +106,10 @@ struct SwrContext { float lfe_mix_level; ///< LFE mixing level float rematrix_volume; ///< rematrixing volume coefficient float rematrix_maxval; ///< maximum value for rematrixing output - enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */ + int matrix_encoding; /**< matrixed stereo encoding */ const int *channel_map; ///< channel index (or -1 if muted channel) map int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) - enum SwrEngine engine; + int engine; struct DitherContext dither; @@ -117,7 +117,7 @@ struct SwrContext { 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 (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */ - enum SwrFilterType filter_type; /**< swr resampling filter type */ + int filter_type; /**< swr resampling filter type */ int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ double precision; /**< soxr resampling precision (in bits) */ int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */ |