audiointerleave.c
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1 /*
2  * Audio Interleaving functions
3  *
4  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
5  *
6  * This file is part of Libav.
7  *
8  * Libav is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * Libav is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with Libav; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include "libavutil/fifo.h"
24 #include "libavutil/mathematics.h"
25 #include "avformat.h"
26 #include "audiointerleave.h"
27 #include "internal.h"
28 
30 {
31  int i;
32  for (i = 0; i < s->nb_streams; i++) {
33  AVStream *st = s->streams[i];
35 
37  av_fifo_free(aic->fifo);
38  }
39 }
40 
42  const int *samples_per_frame,
43  AVRational time_base)
44 {
45  int i;
46 
47  if (!samples_per_frame)
48  return -1;
49 
50  for (i = 0; i < s->nb_streams; i++) {
51  AVStream *st = s->streams[i];
53 
54  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
55  aic->sample_size = (st->codec->channels *
57  if (!aic->sample_size) {
58  av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
59  return -1;
60  }
61  aic->samples_per_frame = samples_per_frame;
62  aic->samples = aic->samples_per_frame;
63  aic->time_base = time_base;
64 
65  aic->fifo_size = 100* *aic->samples;
66  aic->fifo= av_fifo_alloc(100 * *aic->samples);
67  }
68  }
69 
70  return 0;
71 }
72 
74  int stream_index, int flush)
75 {
76  AVStream *st = s->streams[stream_index];
78 
79  int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
80  if (!size || (!flush && size == av_fifo_size(aic->fifo)))
81  return 0;
82 
83  av_new_packet(pkt, size);
84  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
85 
86  pkt->dts = pkt->pts = aic->dts;
87  pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
88  pkt->stream_index = stream_index;
89  aic->dts += pkt->duration;
90 
91  aic->samples++;
92  if (!*aic->samples)
93  aic->samples = aic->samples_per_frame;
94 
95  return size;
96 }
97 
99  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
100  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
101 {
102  int i;
103 
104  if (pkt) {
105  AVStream *st = s->streams[pkt->stream_index];
107  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
108  unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
109  if (new_size > aic->fifo_size) {
110  if (av_fifo_realloc2(aic->fifo, new_size) < 0)
111  return -1;
112  aic->fifo_size = new_size;
113  }
114  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
115  } else {
116  // rewrite pts and dts to be decoded time line position
117  pkt->pts = pkt->dts = aic->dts;
118  aic->dts += pkt->duration;
119  ff_interleave_add_packet(s, pkt, compare_ts);
120  }
121  pkt = NULL;
122  }
123 
124  for (i = 0; i < s->nb_streams; i++) {
125  AVStream *st = s->streams[i];
126  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
127  AVPacket new_pkt;
128  while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
129  ff_interleave_add_packet(s, &new_pkt, compare_ts);
130  }
131  }
132 
133  return get_packet(s, out, pkt, flush);
134 }
int size
int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size, void(*func)(void *, void *, int))
Feed data from an AVFifoBuffer to a user-supplied callback.
Definition: fifo.c:105
int av_fifo_size(AVFifoBuffer *f)
Return the amount of data in bytes in the AVFifoBuffer, that is the amount of data you can read from ...
Definition: fifo.c:52
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:60
const int * samples
current samples per frame, pointer to samples_per_frame
int size
Definition: avcodec.h:909
const int * samples_per_frame
must be 0-terminated
void * priv_data
Definition: avformat.h:633
int av_fifo_realloc2(AVFifoBuffer *f, unsigned int new_size)
Resize an AVFifoBuffer.
Definition: fifo.c:62
Format I/O context.
Definition: avformat.h:863
int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base)
AVStream ** streams
Definition: avformat.h:908
int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size, int(*func)(void *, void *, int))
Feed data from a user-supplied callback to an AVFifoBuffer.
Definition: fifo.c:82
uint8_t * data
Definition: avcodec.h:908
unsigned fifo_size
size of currently allocated FIFO
int av_get_bits_per_sample(enum CodecID codec_id)
Return codec bits per sample.
Definition: utils.c:1634
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:930
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:132
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
Definition: fifo.c:25
static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, int stream_index, int flush)
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, int(*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), int(*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame and interleave them correctly...
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:140
AVCodecContext * codec
codec context
Definition: avformat.h:623
unsigned int nb_streams
A list of all streams in the file.
Definition: avformat.h:907
#define FFMIN(a, b)
Definition: common.h:55
uint64_t dts
current dts
void av_fifo_free(AVFifoBuffer *f)
Free an AVFifoBuffer.
Definition: fifo.c:38
AVRational time_base
time base of output audio packets
Stream structure.
Definition: avformat.h:620
NULL
Definition: eval.c:50
enum AVMediaType codec_type
Definition: avcodec.h:1574
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:111
a very simple circular buffer FIFO implementation
rational number numerator/denominator
Definition: rational.h:43
int sample_size
size of one sample all channels included
Main libavformat public API header.
void ff_audio_interleave_close(AVFormatContext *s)
static av_cold void flush(AVCodecContext *avctx)
Flush (reset) the frame ID after seeking.
Definition: alsdec.c:1779
void ff_interleave_add_packet(AVFormatContext *s, AVPacket *pkt, int(*compare)(AVFormatContext *, AVPacket *, AVPacket *))
Add packet to AVFormatContext->packet_buffer list, determining its interleaved position using compare...
Definition: utils.c:3165
static int get_packet(URLContext *s, int for_header)
Interact with the server by receiving and sending RTMP packets until there is some significant data (...
Definition: rtmpproto.c:680
int channels
number of audio channels
Definition: avcodec.h:1457
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
Definition: avcodec.h:907
int stream_index
Definition: avcodec.h:910
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:652
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:901
enum CodecID codec_id
Definition: avcodec.h:1575