1 /* 2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> 3 * 4 * This file is part of FFmpeg. 5 * 6 * FFmpeg is free software; you can redistribute it and/or 7 * modify it under the terms of the GNU Lesser General Public 8 * License as published by the Free Software Foundation; either 9 * version 2.1 of the License, or (at your option) any later version. 10 * 11 * FFmpeg is distributed in the hope that it will be useful, 12 * but WITHOUT ANY WARRANTY; without even the implied warranty of 13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 14 * Lesser General Public License for more details. 15 * 16 * You should have received a copy of the GNU Lesser General Public 17 * License along with FFmpeg; if not, write to the Free Software 18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 19 */ 20 21 #ifndef AVRESAMPLE_AUDIO_CONVERT_H 22 #define AVRESAMPLE_AUDIO_CONVERT_H 23 24 #include "libavutil/samplefmt.h" 25 #include "avresample.h" 26 #include "internal.h" 27 #include "audio_data.h" 28 29 /** 30 * Set conversion function if the parameters match. 31 * 32 * This compares the parameters of the conversion function to the parameters 33 * in the AudioConvert context. If the parameters do not match, no changes are 34 * made to the active functions. If the parameters do match and the alignment 35 * is not constrained, the function is set as the generic conversion function. 36 * If the parameters match and the alignment is constrained, the function is 37 * set as the optimized conversion function. 38 * 39 * @param ac AudioConvert context 40 * @param out_fmt output sample format 41 * @param in_fmt input sample format 42 * @param channels number of channels, or 0 for any number of channels 43 * @param ptr_align buffer pointer alignment, in bytes 44 * @param samples_align buffer size alignment, in samples 45 * @param descr function type description (e.g. "C" or "SSE") 46 * @param conv conversion function pointer 47 */ 48 void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt, 49 enum AVSampleFormat in_fmt, int channels, 50 int ptr_align, int samples_align, 51 const char *descr, void *conv); 52 53 /** 54 * Allocate and initialize AudioConvert context for sample format conversion. 55 * 56 * @param avr AVAudioResampleContext 57 * @param out_fmt output sample format 58 * @param in_fmt input sample format 59 * @param channels number of channels 60 * @param sample_rate sample rate (used for dithering) 61 * @param apply_map apply channel map during conversion 62 * @return newly-allocated AudioConvert context 63 */ 64 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, 65 enum AVSampleFormat out_fmt, 66 enum AVSampleFormat in_fmt, 67 int channels, int sample_rate, 68 int apply_map); 69 70 /** 71 * Free AudioConvert. 72 * 73 * The AudioConvert must have been previously allocated with ff_audio_convert_alloc(). 74 * 75 * @param ac AudioConvert struct 76 */ 77 void ff_audio_convert_free(AudioConvert **ac); 78 79 /** 80 * Convert audio data from one sample format to another. 81 * 82 * For each call, the alignment of the input and output AudioData buffers are 83 * examined to determine whether to use the generic or optimized conversion 84 * function (when available). 85 * 86 * The number of samples to convert is determined by in->nb_samples. The output 87 * buffer must be large enough to handle this many samples. out->nb_samples is 88 * set by this function before a successful return. 89 * 90 * @param ac AudioConvert context 91 * @param out output audio data 92 * @param in input audio data 93 * @return 0 on success, negative AVERROR code on failure 94 */ 95 int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in); 96 97 /* arch-specific initialization functions */ 98 99 void ff_audio_convert_init_aarch64(AudioConvert *ac); 100 void ff_audio_convert_init_arm(AudioConvert *ac); 101 void ff_audio_convert_init_x86(AudioConvert *ac); 102 103 #endif /* AVRESAMPLE_AUDIO_CONVERT_H */ 104