mirror of https://git.ffmpeg.org/ffmpeg.git
swresample: Use int instead of enum for fields which are accessed through AVOptions as int
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -53,7 +53,7 @@ typedef struct AudioData{
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} AudioData;
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} AudioData;
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struct DitherContext {
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struct DitherContext {
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enum SwrDitherType method;
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int method;
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int noise_pos;
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int noise_pos;
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float scale;
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float scale;
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float noise_scale; ///< Noise scale
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float noise_scale; ///< Noise scale
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@ -106,10 +106,10 @@ struct SwrContext {
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float lfe_mix_level; ///< LFE mixing level
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float lfe_mix_level; ///< LFE mixing level
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float rematrix_volume; ///< rematrixing volume coefficient
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float rematrix_volume; ///< rematrixing volume coefficient
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float rematrix_maxval; ///< maximum value for rematrixing output
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float rematrix_maxval; ///< maximum value for rematrixing output
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enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
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int matrix_encoding; /**< matrixed stereo encoding */
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const int *channel_map; ///< channel index (or -1 if muted channel) map
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const int *channel_map; ///< channel index (or -1 if muted channel) map
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int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
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int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
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enum SwrEngine engine;
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int engine;
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struct DitherContext dither;
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struct DitherContext dither;
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@ -117,7 +117,7 @@ struct SwrContext {
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int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
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int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
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int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
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int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
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double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
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double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
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enum SwrFilterType filter_type; /**< swr resampling filter type */
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int filter_type; /**< swr resampling filter type */
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int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
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int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
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double precision; /**< soxr resampling precision (in bits) */
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double precision; /**< soxr resampling precision (in bits) */
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int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
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int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
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