/* * Copyright (C) 2011 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" 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 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 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 */ 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; 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_2_1_func_type *mix_2_1_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); 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_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt); void swri_audio_convert_init_x86(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels); #endif