ffmpeg/libswresample/swresample_internal.h
Michael Niedermayer 503e5e9db5 libswresample: support directly converting sampleformats and packed/planar in a single pass.
Previously a intermediate planar format was used when both input and output where packed.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-06 00:07:03 +02:00

80 lines
2.8 KiB
C

/*
* 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 struct AudioData{
uint8_t *ch[SWR_CH_MAX];
uint8_t *data;
int ch_count;
int bps;
int count;
int planar;
} AudioData;
typedef struct SwrContext { //FIXME find unused fields
AVClass *av_class;
int log_level_offset;
void *log_ctx;
enum AVSampleFormat in_sample_fmt;
enum AVSampleFormat int_sample_fmt; ///<AV_SAMPLE_FMT_FLT OR AV_SAMPLE_FMT_S16
enum AVSampleFormat out_sample_fmt;
int64_t in_ch_layout;
int64_t out_ch_layout;
int in_sample_rate;
int out_sample_rate;
int flags;
float slev, clev;
//below are private
int int_bps;
int resample_first;
int rematrix; ///< flag to indicate if rematrixing is used
AudioData in, postin, midbuf, preout, out, in_buffer;
int in_buffer_index;
int in_buffer_count;
int resample_in_constraint;
struct AVAudioConvert *in_convert;
struct AVAudioConvert *out_convert;
struct AVAudioConvert *full_convert;
struct AVResampleContext *resample;
float matrix[SWR_CH_MAX][SWR_CH_MAX];
int16_t matrix16[SWR_CH_MAX][SWR_CH_MAX];
uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];
//TODO callbacks for asm optims
}SwrContext;
struct AVResampleContext *swr_resample_init(struct AVResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff);
void swr_resample_free(struct AVResampleContext **c);
int swr_multiple_resample(struct AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
void swr_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
int swr_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
int swr_rematrix_init(SwrContext *s);
int swr_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
#endif