0,0 → 1,103 |
/* |
* 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_AUDIO_CONVERT_H |
#define AVRESAMPLE_AUDIO_CONVERT_H |
|
#include "libavutil/samplefmt.h" |
#include "avresample.h" |
#include "internal.h" |
#include "audio_data.h" |
|
/** |
* Set conversion function if the parameters match. |
* |
* This compares the parameters of the conversion function to the parameters |
* in the AudioConvert context. If the parameters do not match, no changes are |
* made to the active functions. If the parameters do match and the alignment |
* is not constrained, the function is set as the generic conversion function. |
* If the parameters match and the alignment is constrained, the function is |
* set as the optimized conversion function. |
* |
* @param ac AudioConvert context |
* @param out_fmt output sample format |
* @param in_fmt input sample format |
* @param channels number of channels, or 0 for any number of channels |
* @param ptr_align buffer pointer alignment, in bytes |
* @param samples_align buffer size alignment, in samples |
* @param descr function type description (e.g. "C" or "SSE") |
* @param conv conversion function pointer |
*/ |
void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt, |
enum AVSampleFormat in_fmt, int channels, |
int ptr_align, int samples_align, |
const char *descr, void *conv); |
|
/** |
* Allocate and initialize AudioConvert context for sample format conversion. |
* |
* @param avr AVAudioResampleContext |
* @param out_fmt output sample format |
* @param in_fmt input sample format |
* @param channels number of channels |
* @param sample_rate sample rate (used for dithering) |
* @param apply_map apply channel map during conversion |
* @return newly-allocated AudioConvert context |
*/ |
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, |
enum AVSampleFormat out_fmt, |
enum AVSampleFormat in_fmt, |
int channels, int sample_rate, |
int apply_map); |
|
/** |
* Free AudioConvert. |
* |
* The AudioConvert must have been previously allocated with ff_audio_convert_alloc(). |
* |
* @param ac AudioConvert struct |
*/ |
void ff_audio_convert_free(AudioConvert **ac); |
|
/** |
* Convert audio data from one sample format to another. |
* |
* For each call, the alignment of the input and output AudioData buffers are |
* examined to determine whether to use the generic or optimized conversion |
* function (when available). |
* |
* The number of samples to convert is determined by in->nb_samples. The output |
* buffer must be large enough to handle this many samples. out->nb_samples is |
* set by this function before a successful return. |
* |
* @param ac AudioConvert context |
* @param out output audio data |
* @param in input audio data |
* @return 0 on success, negative AVERROR code on failure |
*/ |
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in); |
|
/* arch-specific initialization functions */ |
|
void ff_audio_convert_init_aarch64(AudioConvert *ac); |
void ff_audio_convert_init_arm(AudioConvert *ac); |
void ff_audio_convert_init_x86(AudioConvert *ac); |
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#endif /* AVRESAMPLE_AUDIO_CONVERT_H */ |