FFmpeg  4.0
audio_frame_queue.c
Go to the documentation of this file.
1 /*
2  * Audio Frame Queue
3  * Copyright (c) 2012 Justin Ruggles
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/attributes.h"
23 #include "libavutil/common.h"
24 #include "audio_frame_queue.h"
25 #include "internal.h"
26 #include "libavutil/avassert.h"
27 
29 {
30  afq->avctx = avctx;
31  afq->remaining_delay = avctx->initial_padding;
32  afq->remaining_samples = avctx->initial_padding;
33  afq->frame_count = 0;
34 }
35 
37 {
38  if(afq->frame_count)
39  av_log(afq->avctx, AV_LOG_WARNING, "%d frames left in the queue on closing\n", afq->frame_count);
40  av_freep(&afq->frames);
41  memset(afq, 0, sizeof(*afq));
42 }
43 
45 {
46  AudioFrame *new = av_fast_realloc(afq->frames, &afq->frame_alloc, sizeof(*afq->frames)*(afq->frame_count+1));
47  if(!new)
48  return AVERROR(ENOMEM);
49  afq->frames = new;
50  new += afq->frame_count;
51 
52  /* get frame parameters */
53  new->duration = f->nb_samples;
54  new->duration += afq->remaining_delay;
55  if (f->pts != AV_NOPTS_VALUE) {
56  new->pts = av_rescale_q(f->pts,
57  afq->avctx->time_base,
58  (AVRational){ 1, afq->avctx->sample_rate });
59  new->pts -= afq->remaining_delay;
60  if(afq->frame_count && new[-1].pts >= new->pts)
61  av_log(afq->avctx, AV_LOG_WARNING, "Queue input is backward in time\n");
62  } else {
63  new->pts = AV_NOPTS_VALUE;
64  }
65  afq->remaining_delay = 0;
66 
67  /* add frame sample count */
68  afq->remaining_samples += f->nb_samples;
69 
70  afq->frame_count++;
71 
72  return 0;
73 }
74 
75 void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
76  int64_t *duration)
77 {
78  int64_t out_pts = AV_NOPTS_VALUE;
79  int removed_samples = 0;
80  int i;
81 
82  if (afq->frame_count || afq->frame_alloc) {
83  if (afq->frames->pts != AV_NOPTS_VALUE)
84  out_pts = afq->frames->pts;
85  }
86  if(!afq->frame_count)
87  av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but the queue is empty\n", nb_samples);
88  if (pts)
89  *pts = ff_samples_to_time_base(afq->avctx, out_pts);
90 
91  for(i=0; nb_samples && i<afq->frame_count; i++){
92  int n= FFMIN(afq->frames[i].duration, nb_samples);
93  afq->frames[i].duration -= n;
94  nb_samples -= n;
95  removed_samples += n;
96  if(afq->frames[i].pts != AV_NOPTS_VALUE)
97  afq->frames[i].pts += n;
98  }
99  afq->remaining_samples -= removed_samples;
100  i -= i && afq->frames[i-1].duration;
101  memmove(afq->frames, afq->frames + i, sizeof(*afq->frames) * (afq->frame_count - i));
102  afq->frame_count -= i;
103 
104  if(nb_samples){
105  av_assert0(!afq->frame_count);
107  if(afq->frames && afq->frames[0].pts != AV_NOPTS_VALUE)
108  afq->frames[0].pts += nb_samples;
109  av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than there are in the queue\n", nb_samples);
110  }
111  if (duration)
112  *duration = ff_samples_to_time_base(afq->avctx, removed_samples);
113 }
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
AudioFrame * frames
Macro definitions for various function/variable attributes.
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avcodec.h:1640
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
#define av_cold
Definition: attributes.h:82
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:311
int64_t duration
Definition: movenc.c:63
#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
#define AVERROR(e)
Definition: error.h:43
int initial_padding
Audio only.
Definition: avcodec.h:3031
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
simple assert() macros that are a bit more flexible than ISO C assert().
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
AVCodecContext * avctx
#define FFMIN(a, b)
Definition: common.h:96
void * av_fast_realloc(void *ptr, unsigned int *size, size_t min_size)
Reallocate the given buffer if it is not large enough, otherwise do nothing.
Definition: mem.c:464
int n
Definition: avisynth_c.h:684
main external API structure.
Definition: avcodec.h:1518
Rational number (pair of numerator and denominator).
Definition: rational.h:58
static int64_t pts
common internal api header.
common internal and external API header
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define av_freep(p)
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
Definition: internal.h:280
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248