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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 FFmpeg.
7  *
8  * FFmpeg 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  * FFmpeg 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 FFmpeg; 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_freep(&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 AVERROR(EINVAL);
49 
50  if (!time_base.num) {
51  av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
52  return AVERROR(EINVAL);
53  }
54  for (i = 0; i < s->nb_streams; i++) {
55  AVStream *st = s->streams[i];
57 
59  aic->sample_size = (st->codecpar->channels *
61  if (!aic->sample_size) {
62  av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
63  return AVERROR(EINVAL);
64  }
65  aic->samples_per_frame = samples_per_frame;
66  aic->samples = aic->samples_per_frame;
67  aic->time_base = time_base;
68 
69  aic->fifo_size = 100* *aic->samples;
70  if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
71  return AVERROR(ENOMEM);
72  }
73  }
74 
75  return 0;
76 }
77 
79  int stream_index, int flush)
80 {
81  AVStream *st = s->streams[stream_index];
83  int ret;
84  int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
85  if (!size || (!flush && size == av_fifo_size(aic->fifo)))
86  return 0;
87 
88  ret = av_new_packet(pkt, size);
89  if (ret < 0)
90  return ret;
91  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
92 
93  pkt->dts = pkt->pts = aic->dts;
94  pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
95  pkt->stream_index = stream_index;
96  aic->dts += pkt->duration;
97 
98  aic->samples++;
99  if (!*aic->samples)
100  aic->samples = aic->samples_per_frame;
101 
102  return size;
103 }
104 
106  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
107  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
108 {
109  int i, ret;
110 
111  if (pkt) {
112  AVStream *st = s->streams[pkt->stream_index];
114  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
115  unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
116  if (new_size > aic->fifo_size) {
117  if (av_fifo_realloc2(aic->fifo, new_size) < 0)
118  return AVERROR(ENOMEM);
119  aic->fifo_size = new_size;
120  }
121  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
122  } else {
123  // rewrite pts and dts to be decoded time line position
124  pkt->pts = pkt->dts = aic->dts;
125  aic->dts += pkt->duration;
126  if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
127  return ret;
128  }
129  pkt = NULL;
130  }
131 
132  for (i = 0; i < s->nb_streams; i++) {
133  AVStream *st = s->streams[i];
134  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
135  AVPacket new_pkt = { 0 };
136  while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
137  if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
138  return ret;
139  }
140  if (ret < 0)
141  return ret;
142  }
143  }
144 
145  return get_packet(s, out, NULL, flush);
146 }
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:631
static void flush(AVCodecContext *avctx)
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:3939
const int * samples
current samples per frame, pointer to samples_per_frame
int num
numerator
Definition: rational.h:44
int size
Definition: avcodec.h:1589
const int * samples_per_frame
must be 0-terminated
void * priv_data
Definition: avformat.h:891
static AVPacket pkt
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:122
Format I/O context.
Definition: avformat.h:1325
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1606
int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base)
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1393
uint8_t * data
Definition: avcodec.h:1588
unsigned fifo_size
size of currently allocated FIFO
ptrdiff_t size
Definition: opengl_enc.c:101
#define av_log(a,...)
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:86
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:3446
#define AVERROR(e)
Definition: error.h:43
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...
int 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: mux.c:918
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:213
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:3935
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1381
#define FFMIN(a, b)
Definition: common.h:96
uint64_t dts
current dts
AVRational time_base
time base of output audio packets
Stream structure.
Definition: avformat.h:876
int av_fifo_size(const 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:77
int av_fifo_realloc2(AVFifoBuffer *f, unsigned int new_size)
Resize an AVFifoBuffer.
Definition: fifo.c:87
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
AVFifoBuffer * av_fifo_alloc_array(size_t nmemb, size_t size)
Initialize an AVFifoBuffer.
Definition: fifo.c:49
Main libavformat public API header.
void ff_audio_interleave_close(AVFormatContext *s)
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:2396
static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, int stream_index, int flush)
int channels
Audio only.
Definition: avcodec.h:4045
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
Definition: avcodec.h:1587
FILE * out
Definition: movenc.c:54
void av_fifo_freep(AVFifoBuffer **f)
Free an AVFifoBuffer and reset pointer to NULL.
Definition: fifo.c:63
AVCodecParameters * codecpar
Definition: avformat.h:1228
int stream_index
Definition: avcodec.h:1590
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:913
This structure stores compressed data.
Definition: avcodec.h:1565
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1581