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- /*
- * 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 */
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