mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-18 15:25:05 +02:00
![Michael Niedermayer](/assets/img/avatar_default.png)
* commit 'a24a252709dd38f12aa4929ce4981f87091a5113': aarch64: NEON optimized FIR audio resampling Merged-by: Michael Niedermayer <michaelni@gmx.at>
114 lines
5.7 KiB
C
114 lines
5.7 KiB
C
/*
|
|
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg 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.
|
|
*
|
|
* FFmpeg 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 FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
#ifndef AVRESAMPLE_INTERNAL_H
|
|
#define AVRESAMPLE_INTERNAL_H
|
|
|
|
#include "libavutil/audio_fifo.h"
|
|
#include "libavutil/log.h"
|
|
#include "libavutil/opt.h"
|
|
#include "libavutil/samplefmt.h"
|
|
#include "avresample.h"
|
|
|
|
typedef struct AudioData AudioData;
|
|
typedef struct AudioConvert AudioConvert;
|
|
typedef struct AudioMix AudioMix;
|
|
typedef struct ResampleContext ResampleContext;
|
|
|
|
enum RemapPoint {
|
|
REMAP_NONE,
|
|
REMAP_IN_COPY,
|
|
REMAP_IN_CONVERT,
|
|
REMAP_OUT_COPY,
|
|
REMAP_OUT_CONVERT,
|
|
};
|
|
|
|
typedef struct ChannelMapInfo {
|
|
int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */
|
|
int do_remap; /**< remap needed */
|
|
int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */
|
|
int do_copy; /**< copy needed */
|
|
int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */
|
|
int do_zero; /**< zeroing needed */
|
|
int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */
|
|
} ChannelMapInfo;
|
|
|
|
struct AVAudioResampleContext {
|
|
const AVClass *av_class; /**< AVClass for logging and AVOptions */
|
|
|
|
uint64_t in_channel_layout; /**< input channel layout */
|
|
enum AVSampleFormat in_sample_fmt; /**< input sample format */
|
|
int in_sample_rate; /**< input sample rate */
|
|
uint64_t out_channel_layout; /**< output channel layout */
|
|
enum AVSampleFormat out_sample_fmt; /**< output sample format */
|
|
int out_sample_rate; /**< output sample rate */
|
|
enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
|
|
enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
|
|
double center_mix_level; /**< center mix level */
|
|
double surround_mix_level; /**< surround mix level */
|
|
double lfe_mix_level; /**< lfe mix level */
|
|
int normalize_mix_level; /**< enable mix level normalization */
|
|
int force_resampling; /**< force resampling */
|
|
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 AVResampleFilterType filter_type; /**< resampling filter type */
|
|
int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
|
|
enum AVResampleDitherMethod dither_method; /**< dither method */
|
|
|
|
int in_channels; /**< number of input channels */
|
|
int out_channels; /**< number of output channels */
|
|
int resample_channels; /**< number of channels used for resampling */
|
|
int downmix_needed; /**< downmixing is needed */
|
|
int upmix_needed; /**< upmixing is needed */
|
|
int mixing_needed; /**< either upmixing or downmixing is needed */
|
|
int resample_needed; /**< resampling is needed */
|
|
int in_convert_needed; /**< input sample format conversion is needed */
|
|
int out_convert_needed; /**< output sample format conversion is needed */
|
|
int in_copy_needed; /**< input data copy is needed */
|
|
|
|
AudioData *in_buffer; /**< buffer for converted input */
|
|
AudioData *resample_out_buffer; /**< buffer for output from resampler */
|
|
AudioData *out_buffer; /**< buffer for converted output */
|
|
AVAudioFifo *out_fifo; /**< FIFO for output samples */
|
|
|
|
AudioConvert *ac_in; /**< input sample format conversion context */
|
|
AudioConvert *ac_out; /**< output sample format conversion context */
|
|
ResampleContext *resample; /**< resampling context */
|
|
AudioMix *am; /**< channel mixing context */
|
|
enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
|
|
|
|
/**
|
|
* mix matrix
|
|
* only used if avresample_set_matrix() is called before avresample_open()
|
|
*/
|
|
double *mix_matrix;
|
|
|
|
int use_channel_map;
|
|
enum RemapPoint remap_point;
|
|
ChannelMapInfo ch_map_info;
|
|
};
|
|
|
|
|
|
void ff_audio_resample_init_aarch64(ResampleContext *c,
|
|
enum AVSampleFormat sample_fmt);
|
|
#endif /* AVRESAMPLE_INTERNAL_H */
|