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Diffstat (limited to 'libswresample/swresample_internal.h')
| -rw-r--r-- | libswresample/swresample_internal.h | 142 |
1 files changed, 142 insertions, 0 deletions
diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h new file mode 100644 index 0000000000..ab17602c64 --- /dev/null +++ b/libswresample/swresample_internal.h @@ -0,0 +1,142 @@ +/* + * Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at) + * + * This file is part of libswresample + * + * libswresample is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * libswresample is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with libswresample; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef SWR_INTERNAL_H +#define SWR_INTERNAL_H + +#include "swresample.h" +#include "libavutil/audioconvert.h" + +#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */ + +typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, int index, int len); +typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, int index1, int index2, int len); + +typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, int len); + +typedef struct AudioData{ + uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel + uint8_t *data; ///< samples buffer + int ch_count; ///< number of channels + int bps; ///< bytes per sample + int count; ///< number of samples + int planar; ///< 1 if planar audio, 0 otherwise + enum AVSampleFormat fmt; ///< sample format +} AudioData; + +struct SwrContext { + const AVClass *av_class; ///< AVClass used for AVOption and av_log() + int log_level_offset; ///< logging level offset + void *log_ctx; ///< parent logging context + enum AVSampleFormat in_sample_fmt; ///< input sample format + enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P) + enum AVSampleFormat out_sample_fmt; ///< output sample format + int64_t in_ch_layout; ///< input channel layout + int64_t out_ch_layout; ///< output channel layout + int in_sample_rate; ///< input sample rate + int out_sample_rate; ///< output sample rate + int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE + float slev; ///< surround mixing level + float clev; ///< center mixing level + float lfe_mix_level; ///< LFE mixing level + float rematrix_volume; ///< rematrixing volume coefficient + enum AVMatrixEncoding 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 SwrDitherType dither_method; + int dither_pos; + float dither_scale; + int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */ + 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 SwrFilterType filter_type; /**< resampling filter type */ + int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ + + float min_compensation; ///< minimum below which no compensation will happen + float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen + float soft_compensation_duration; ///< duration over which soft compensation is applied + float max_soft_compensation; ///< maximum soft compensation in seconds over soft_compensation_duration + + int resample_first; ///< 1 if resampling must come first, 0 if rematrixing + int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch) + int rematrix_custom; ///< flag to indicate that a custom matrix has been defined + + AudioData in; ///< input audio data + AudioData postin; ///< post-input audio data: used for rematrix/resample + AudioData midbuf; ///< intermediate audio data (postin/preout) + AudioData preout; ///< pre-output audio data: used for rematrix/resample + AudioData out; ///< converted output audio data + AudioData in_buffer; ///< cached audio data (convert and resample purpose) + AudioData dither; ///< noise used for dithering + int in_buffer_index; ///< cached buffer position + int in_buffer_count; ///< cached buffer length + int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise + int flushed; ///< 1 if data is to be flushed and no further input is expected + int64_t outpts; ///< output PTS + int drop_output; ///< number of output samples to drop + + struct AudioConvert *in_convert; ///< input conversion context + struct AudioConvert *out_convert; ///< output conversion context + struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output) + struct ResampleContext *resample; ///< resampling context + + float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients + uint8_t *native_matrix; + uint8_t *native_one; + uint8_t *native_simd_matrix; + int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients + uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients + mix_1_1_func_type *mix_1_1_f; + mix_1_1_func_type *mix_1_1_simd; + + mix_2_1_func_type *mix_2_1_f; + mix_2_1_func_type *mix_2_1_simd; + + mix_any_func_type *mix_any_f; + + /* TODO: callbacks for ASM optimizations */ +}; + +struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat, enum SwrFilterType, int kaiser_beta); +void swri_resample_free(struct ResampleContext **c); +int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); +void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance); +int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx); +int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx); +int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx); +int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx); + +int swri_rematrix_init(SwrContext *s); +void swri_rematrix_free(SwrContext *s); +int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy); +void swri_rematrix_init_x86(struct SwrContext *s); + +void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt); + +void swri_audio_convert_init_arm(struct AudioConvert *ac, + enum AVSampleFormat out_fmt, + enum AVSampleFormat in_fmt, + int channels); +void swri_audio_convert_init_x86(struct AudioConvert *ac, + enum AVSampleFormat out_fmt, + enum AVSampleFormat in_fmt, + int channels); +#endif |