mirror of https://github.com/FFmpeg/FFmpeg.git
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
132 lines
8.3 KiB
132 lines
8.3 KiB
/* |
|
* 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 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 |
|
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; |
|
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); |
|
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_x86(struct AudioConvert *ac, |
|
enum AVSampleFormat out_fmt, |
|
enum AVSampleFormat in_fmt, |
|
int channels); |
|
#endif
|
|
|