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Diffstat (limited to 'libavresample/audio_convert.h')
| -rw-r--r-- | libavresample/audio_convert.h | 103 |
1 files changed, 0 insertions, 103 deletions
diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h deleted file mode 100644 index df15442c18..0000000000 --- a/libavresample/audio_convert.h +++ /dev/null @@ -1,103 +0,0 @@ -/* - * 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); - -#endif /* AVRESAMPLE_AUDIO_CONVERT_H */ |