lavr: typedef internal structs in internal.h
[libav.git] / libavresample / audio_convert.h
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"
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 * @return newly-allocated AudioConvert context
62 */
63 AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
64 enum AVSampleFormat out_fmt,
65 enum AVSampleFormat in_fmt,
66 int channels, int sample_rate);
67
68 /**
69 * Free AudioConvert.
70 *
71 * The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
72 *
73 * @param ac AudioConvert struct
74 */
75 void ff_audio_convert_free(AudioConvert **ac);
76
77 /**
78 * Convert audio data from one sample format to another.
79 *
80 * For each call, the alignment of the input and output AudioData buffers are
81 * examined to determine whether to use the generic or optimized conversion
82 * function (when available).
83 *
84 * The number of samples to convert is determined by in->nb_samples. The output
85 * buffer must be large enough to handle this many samples. out->nb_samples is
86 * set by this function before a successful return.
87 *
88 * @param ac AudioConvert context
89 * @param out output audio data
90 * @param in input audio data
91 * @return 0 on success, negative AVERROR code on failure
92 */
93 int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
94
95 /* arch-specific initialization functions */
96
97 void ff_audio_convert_init_arm(AudioConvert *ac);
98 void ff_audio_convert_init_x86(AudioConvert *ac);
99
100 #endif /* AVRESAMPLE_AUDIO_CONVERT_H */