Libav
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  int ret;
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  ret = av_new_packet(pkt, size);
84  if (ret < 0)
85  return ret;
86  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
87 
88  pkt->dts = pkt->pts = aic->dts;
89  pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
90  pkt->stream_index = stream_index;
91  aic->dts += pkt->duration;
92 
93  aic->samples++;
94  if (!*aic->samples)
95  aic->samples = aic->samples_per_frame;
96 
97  return size;
98 }
99 
101  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
102  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
103 {
104  int i, ret;
105 
106  if (pkt) {
107  AVStream *st = s->streams[pkt->stream_index];
109  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
110  unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
111  if (new_size > aic->fifo_size) {
112  if (av_fifo_realloc2(aic->fifo, new_size) < 0)
113  return -1;
114  aic->fifo_size = new_size;
115  }
116  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
117  } else {
118  // rewrite pts and dts to be decoded time line position
119  pkt->pts = pkt->dts = aic->dts;
120  aic->dts += pkt->duration;
121  if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
122  return ret;
123  }
124  pkt = NULL;
125  }
126 
127  for (i = 0; i < s->nb_streams; i++) {
128  AVStream *st = s->streams[i];
129  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
130  AVPacket new_pkt = { 0 };
131  while (interleave_new_audio_packet(s, &new_pkt, i, flush))
132  if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
133  return ret;
134  }
135  }
136 
137  return get_packet(s, out, NULL, flush);
138 }
int size
const int * samples
current samples per frame, pointer to samples_per_frame
int size
Definition: avcodec.h:974
const int * samples_per_frame
must be 0-terminated
void * priv_data
Definition: avformat.h:719
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:84
Format I/O context.
Definition: avformat.h:922
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:990
uint8_t * data
Definition: avcodec.h:973
unsigned fifo_size
size of currently allocated FIFO
void av_fifo_free(AVFifoBuffer *f)
Free an AVFifoBuffer.
Definition: fifo.c:38
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:991
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:129
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:81
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:2043
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:391
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:107
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:169
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:718
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:978
#define FFMIN(a, b)
Definition: common.h:57
uint64_t dts
current dts
AVRational time_base
time base of output audio packets
Stream structure.
Definition: avformat.h:699
NULL
Definition: eval.c:55
enum AVMediaType codec_type
Definition: avcodec.h:1058
enum AVCodecID codec_id
Definition: avcodec.h:1067
int av_fifo_realloc2(AVFifoBuffer *f, unsigned int new_size)
Resize an AVFifoBuffer.
Definition: fifo.c:62
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
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
Main libavformat public API header.
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
void ff_audio_interleave_close(AVFormatContext *s)
static av_cold void flush(AVCodecContext *avctx)
Flush (reset) the frame ID after seeking.
Definition: alsdec.c:1797
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:2315
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
Definition: fifo.c:25
static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, int stream_index, int flush)
int channels
number of audio channels
Definition: avcodec.h:1808
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
Definition: avcodec.h:972
int stream_index
Definition: avcodec.h:975
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:741
This structure stores compressed data.
Definition: avcodec.h:950
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:966