FFmpeg
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 frame_size = *aic->samples * aic->sample_size;
85  int size = FFMIN(av_fifo_size(aic->fifo), frame_size);
86  if (!size || (!flush && size == av_fifo_size(aic->fifo)))
87  return 0;
88 
89  ret = av_new_packet(pkt, frame_size);
90  if (ret < 0)
91  return ret;
92  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
93 
94  if (size < pkt->size)
95  memset(pkt->data + size, 0, pkt->size - size);
96 
97  pkt->dts = pkt->pts = aic->dts;
98  pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
99  pkt->stream_index = stream_index;
100  aic->dts += pkt->duration;
101 
102  aic->samples++;
103  if (!*aic->samples)
104  aic->samples = aic->samples_per_frame;
105 
106  return pkt->size;
107 }
108 
110  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
111  int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *))
112 {
113  int i, ret;
114 
115  if (pkt) {
116  AVStream *st = s->streams[pkt->stream_index];
118  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
119  unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
120  if (new_size > aic->fifo_size) {
121  if (av_fifo_realloc2(aic->fifo, new_size) < 0)
122  return AVERROR(ENOMEM);
123  aic->fifo_size = new_size;
124  }
125  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
126  } else {
127  // rewrite pts and dts to be decoded time line position
128  pkt->pts = pkt->dts = aic->dts;
129  aic->dts += pkt->duration;
130  if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
131  return ret;
132  }
133  pkt = NULL;
134  }
135 
136  for (i = 0; i < s->nb_streams; i++) {
137  AVStream *st = s->streams[i];
138  if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
139  AVPacket new_pkt = { 0 };
140  while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
141  if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
142  return ret;
143  }
144  if (ret < 0)
145  return ret;
146  }
147  }
148 
149  return get_packet(s, out, NULL, flush);
150 }
#define NULL
Definition: coverity.c:32
static void flush(AVCodecContext *avctx)
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:3970
const int * samples
current samples per frame, pointer to samples_per_frame
int num
Numerator.
Definition: rational.h:59
int size
Definition: avcodec.h:1483
const int * samples_per_frame
must be 0-terminated
void * priv_data
Definition: avformat.h:896
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:1358
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1500
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:1426
uint8_t * data
Definition: avcodec.h:1482
unsigned fifo_size
size of currently allocated FIFO
ptrdiff_t size
Definition: opengl_enc.c:100
#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 i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#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:1533
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:3966
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1414
#define FFMIN(a, b)
Definition: common.h:96
uint64_t dts
current dts
#define s(width, name)
Definition: cbs_vp9.c:257
AVRational time_base
time base of output audio packets
Stream structure.
Definition: avformat.h:881
int frame_size
Definition: mxfenc.c:2150
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 (pair of numerator and denominator).
Definition: rational.h:58
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)
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, int(*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), int(*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *))
Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame and interleave them correctly...
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:2407
static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, int stream_index, int flush)
int channels
Audio only.
Definition: avcodec.h:4076
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
Definition: avcodec.h:1481
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
Codec parameters associated with this stream.
Definition: avformat.h:1028
int stream_index
Definition: avcodec.h:1484
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:910
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
This structure stores compressed data.
Definition: avcodec.h:1459
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1475
int ff_interleave_add_packet(AVFormatContext *s, AVPacket *pkt, int(*compare)(AVFormatContext *, const AVPacket *, const AVPacket *))
Add packet to AVFormatContext->packet_buffer list, determining its interleaved position using compare...
Definition: mux.c:926