2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of Libav.
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.
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.
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
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"
32 * Audio buffer used for intermediate storage between conversion phases.
34 typedef struct AudioData {
35 const AVClass *class; /**< AVClass for logging */
36 uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
37 uint8_t *buffer; /**< data buffer */
38 unsigned int buffer_size; /**< allocated buffer size */
39 int allocated_samples; /**< number of samples the buffer can hold */
40 int nb_samples; /**< current number of samples */
41 enum AVSampleFormat sample_fmt; /**< sample format */
42 int channels; /**< channel count */
43 int allocated_channels; /**< allocated channel count */
44 int is_planar; /**< sample format is planar */
45 int planes; /**< number of data planes */
46 int sample_size; /**< bytes per sample */
47 int stride; /**< sample byte offset within a plane */
48 int read_only; /**< data is read-only */
49 int allow_realloc; /**< realloc is allowed */
50 int ptr_align; /**< minimum data pointer alignment */
51 int samples_align; /**< allocated samples alignment */
52 const char *name; /**< name for debug logging */
55 int ff_audio_data_set_channels(AudioData *a, int channels);
58 * Initialize AudioData using a given source.
60 * This does not allocate an internal buffer. It only sets the data pointers
61 * and audio parameters.
63 * @param a AudioData struct
64 * @param src source data pointers
65 * @param plane_size plane size, in bytes.
66 * This can be 0 if unknown, but that will lead to
67 * optimized functions not being used in many cases,
68 * which could slow down some conversions.
69 * @param channels channel count
70 * @param nb_samples number of samples in the source data
71 * @param sample_fmt sample format
72 * @param read_only indicates if buffer is read only or read/write
73 * @param name name for debug logging (can be NULL)
74 * @return 0 on success, negative AVERROR value on error
76 int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
77 int nb_samples, enum AVSampleFormat sample_fmt,
78 int read_only, const char *name);
83 * This allocates an internal buffer and sets audio parameters.
85 * @param channels channel count
86 * @param nb_samples number of samples to allocate space for
87 * @param sample_fmt sample format
88 * @param name name for debug logging (can be NULL)
89 * @return newly allocated AudioData struct, or NULL on error
91 AudioData *ff_audio_data_alloc(int channels, int nb_samples,
92 enum AVSampleFormat sample_fmt,
96 * Reallocate AudioData.
98 * The AudioData must have been previously allocated with ff_audio_data_alloc().
100 * @param a AudioData struct
101 * @param nb_samples number of samples to allocate space for
102 * @return 0 on success, negative AVERROR value on error
104 int ff_audio_data_realloc(AudioData *a, int nb_samples);
109 * The AudioData must have been previously allocated with ff_audio_data_alloc().
111 * @param a AudioData struct
113 void ff_audio_data_free(AudioData **a);
116 * Copy data from one AudioData to another.
118 * @param out output AudioData
119 * @param in input AudioData
120 * @return 0 on success, negative AVERROR value on error
122 int ff_audio_data_copy(AudioData *out, AudioData *in);
125 * Append data from one AudioData to the end of another.
127 * @param dst destination AudioData
128 * @param dst_offset offset, in samples, to start writing, relative to the
130 * @param src source AudioData
131 * @param src_offset offset, in samples, to start copying, relative to the
133 * @param nb_samples number of samples to copy
134 * @return 0 on success, negative AVERROR value on error
136 int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
137 int src_offset, int nb_samples);
140 * Drain samples from the start of the AudioData.
142 * Remaining samples are shifted to the start of the AudioData.
144 * @param a AudioData struct
145 * @param nb_samples number of samples to drain
147 void ff_audio_data_drain(AudioData *a, int nb_samples);
150 * Add samples in AudioData to an AVAudioFifo.
152 * @param af Audio FIFO Buffer
153 * @param a AudioData struct
154 * @param offset number of samples to skip from the start of the data
155 * @param nb_samples number of samples to add to the FIFO
156 * @return number of samples actually added to the FIFO, or
157 * negative AVERROR code on error
159 int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
163 * Read samples from an AVAudioFifo to AudioData.
165 * @param af Audio FIFO Buffer
166 * @param a AudioData struct
167 * @param nb_samples number of samples to read from the FIFO
168 * @return number of samples actually read from the FIFO, or
169 * negative AVERROR code on error
171 int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
173 #endif /* AVRESAMPLE_AUDIO_DATA_H */