2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of FFmpeg.
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.
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.
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
21 #ifndef AVRESAMPLE_AUDIO_DATA_H
22 #define AVRESAMPLE_AUDIO_DATA_H
26 #include "libavutil/audio_fifo.h"
27 #include "libavutil/log.h"
28 #include "libavutil/samplefmt.h"
29 #include "avresample.h"
33 * Audio buffer used for intermediate storage between conversion phases.
36 const AVClass *class; /**< AVClass for logging */
37 uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
38 uint8_t *buffer; /**< data buffer */
39 unsigned int buffer_size; /**< allocated buffer size */
40 int allocated_samples; /**< number of samples the buffer can hold */
41 int nb_samples; /**< current number of samples */
42 enum AVSampleFormat sample_fmt; /**< sample format */
43 int channels; /**< channel count */
44 int allocated_channels; /**< allocated channel count */
45 int is_planar; /**< sample format is planar */
46 int planes; /**< number of data planes */
47 int sample_size; /**< bytes per sample */
48 int stride; /**< sample byte offset within a plane */
49 int read_only; /**< data is read-only */
50 int allow_realloc; /**< realloc is allowed */
51 int ptr_align; /**< minimum data pointer alignment */
52 int samples_align; /**< allocated samples alignment */
53 const char *name; /**< name for debug logging */
56 int ff_audio_data_set_channels(AudioData *a, int channels);
59 * Initialize AudioData using a given source.
61 * This does not allocate an internal buffer. It only sets the data pointers
62 * and audio parameters.
64 * @param a AudioData struct
65 * @param src source data pointers
66 * @param plane_size plane size, in bytes.
67 * This can be 0 if unknown, but that will lead to
68 * optimized functions not being used in many cases,
69 * which could slow down some conversions.
70 * @param channels channel count
71 * @param nb_samples number of samples in the source data
72 * @param sample_fmt sample format
73 * @param read_only indicates if buffer is read only or read/write
74 * @param name name for debug logging (can be NULL)
75 * @return 0 on success, negative AVERROR value on error
77 int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
78 int nb_samples, enum AVSampleFormat sample_fmt,
79 int read_only, const char *name);
84 * This allocates an internal buffer and sets audio parameters.
86 * @param channels channel count
87 * @param nb_samples number of samples to allocate space for
88 * @param sample_fmt sample format
89 * @param name name for debug logging (can be NULL)
90 * @return newly allocated AudioData struct, or NULL on error
92 AudioData *ff_audio_data_alloc(int channels, int nb_samples,
93 enum AVSampleFormat sample_fmt,
97 * Reallocate AudioData.
99 * The AudioData must have been previously allocated with ff_audio_data_alloc().
101 * @param a AudioData struct
102 * @param nb_samples number of samples to allocate space for
103 * @return 0 on success, negative AVERROR value on error
105 int ff_audio_data_realloc(AudioData *a, int nb_samples);
110 * The AudioData must have been previously allocated with ff_audio_data_alloc().
112 * @param a AudioData struct
114 void ff_audio_data_free(AudioData **a);
117 * Copy data from one AudioData to another.
119 * @param out output AudioData
120 * @param in input AudioData
121 * @param map channel map, NULL if not remapping
122 * @return 0 on success, negative AVERROR value on error
124 int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map);
127 * Append data from one AudioData to the end of another.
129 * @param dst destination AudioData
130 * @param dst_offset offset, in samples, to start writing, relative to the
132 * @param src source AudioData
133 * @param src_offset offset, in samples, to start copying, relative to the
135 * @param nb_samples number of samples to copy
136 * @return 0 on success, negative AVERROR value on error
138 int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
139 int src_offset, int nb_samples);
142 * Drain samples from the start of the AudioData.
144 * Remaining samples are shifted to the start of the AudioData.
146 * @param a AudioData struct
147 * @param nb_samples number of samples to drain
149 void ff_audio_data_drain(AudioData *a, int nb_samples);
152 * Add samples in AudioData to an AVAudioFifo.
154 * @param af Audio FIFO Buffer
155 * @param a AudioData struct
156 * @param offset number of samples to skip from the start of the data
157 * @param nb_samples number of samples to add to the FIFO
158 * @return number of samples actually added to the FIFO, or
159 * negative AVERROR code on error
161 int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
165 * Read samples from an AVAudioFifo to AudioData.
167 * @param af Audio FIFO Buffer
168 * @param a AudioData struct
169 * @param nb_samples number of samples to read from the FIFO
170 * @return number of samples actually read from the FIFO, or
171 * negative AVERROR code on error
173 int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
175 #endif /* AVRESAMPLE_AUDIO_DATA_H */