x86: lavr: add SSE2/AVX dither_int_to_float()
[libav.git] / libavresample / audio_convert.h
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1/*
2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3 *
4 * This file is part of Libav.
5 *
6 * Libav 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 * Libav 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 Libav; 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"
4d68269d 26#include "internal.h"
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27#include "audio_data.h"
28
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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
804d7a1a 44 * @param samples_align buffer size alignment, in samples
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45 * @param descr function type description (e.g. "C" or "SSE")
46 * @param conv conversion function pointer
47 */
48void 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 *
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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)
074a00d1 61 * @param apply_map apply channel map during conversion
b2fe6756 62 * @return newly-allocated AudioConvert context
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63 */
64AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
65 enum AVSampleFormat out_fmt,
66 enum AVSampleFormat in_fmt,
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67 int channels, int sample_rate,
68 int apply_map);
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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 */
77void ff_audio_convert_free(AudioConvert **ac);
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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 *
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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 *
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90 * @param ac AudioConvert context
91 * @param out output audio data
92 * @param in input audio data
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93 * @return 0 on success, negative AVERROR code on failure
94 */
7f534d11 95int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
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96
97/* arch-specific initialization functions */
98
643b5b79 99void ff_audio_convert_init_arm(AudioConvert *ac);
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100void ff_audio_convert_init_x86(AudioConvert *ac);
101
102#endif /* AVRESAMPLE_AUDIO_CONVERT_H */