1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
|
/*
* Audio Frame Queue
* Copyright (c) 2012 Justin Ruggles
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/common.h"
#include "libavutil/mathematics.h"
#include "internal.h"
#include "audio_frame_queue.h"
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
{
afq->avctx = avctx;
afq->next_pts = AV_NOPTS_VALUE;
afq->remaining_delay = avctx->delay;
afq->remaining_samples = avctx->delay;
afq->frame_queue = NULL;
}
static void delete_next_frame(AudioFrameQueue *afq)
{
AudioFrame *f = afq->frame_queue;
if (f) {
afq->frame_queue = f->next;
f->next = NULL;
av_freep(&f);
}
}
void ff_af_queue_close(AudioFrameQueue *afq)
{
/* remove/free any remaining frames */
while (afq->frame_queue)
delete_next_frame(afq);
memset(afq, 0, sizeof(*afq));
}
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
{
AudioFrame *new_frame;
AudioFrame *queue_end = afq->frame_queue;
/* find the end of the queue */
while (queue_end && queue_end->next)
queue_end = queue_end->next;
/* allocate new frame queue entry */
if (!(new_frame = av_malloc(sizeof(*new_frame))))
return AVERROR(ENOMEM);
/* get frame parameters */
new_frame->next = NULL;
new_frame->duration = f->nb_samples;
if (f->pts != AV_NOPTS_VALUE) {
new_frame->pts = av_rescale_q(f->pts,
afq->avctx->time_base,
(AVRational){ 1, afq->avctx->sample_rate });
afq->next_pts = new_frame->pts + new_frame->duration;
} else {
new_frame->pts = AV_NOPTS_VALUE;
afq->next_pts = AV_NOPTS_VALUE;
}
/* add new frame to the end of the queue */
if (!queue_end)
afq->frame_queue = new_frame;
else
queue_end->next = new_frame;
/* add frame sample count */
afq->remaining_samples += f->nb_samples;
#ifdef DEBUG
ff_af_queue_log_state(afq);
#endif
return 0;
}
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts,
int *duration)
{
int64_t out_pts = AV_NOPTS_VALUE;
int removed_samples = 0;
#ifdef DEBUG
ff_af_queue_log_state(afq);
#endif
/* get output pts from the next frame or generated pts */
if (afq->frame_queue) {
if (afq->frame_queue->pts != AV_NOPTS_VALUE)
out_pts = afq->frame_queue->pts - afq->remaining_delay;
} else {
if (afq->next_pts != AV_NOPTS_VALUE)
out_pts = afq->next_pts - afq->remaining_delay;
}
if (pts) {
if (out_pts != AV_NOPTS_VALUE)
*pts = ff_samples_to_time_base(afq->avctx, out_pts);
else
*pts = AV_NOPTS_VALUE;
}
/* if the delay is larger than the packet duration, we use up delay samples
for the output packet and leave all frames in the queue */
if (afq->remaining_delay >= nb_samples) {
removed_samples += nb_samples;
afq->remaining_delay -= nb_samples;
}
/* remove frames from the queue until we have enough to cover the
requested number of samples or until the queue is empty */
while (removed_samples < nb_samples && afq->frame_queue) {
removed_samples += afq->frame_queue->duration;
delete_next_frame(afq);
}
afq->remaining_samples -= removed_samples;
/* if there are no frames left and we have room for more samples, use
any remaining delay samples */
if (removed_samples < nb_samples && afq->remaining_samples > 0) {
int add_samples = FFMIN(afq->remaining_samples,
nb_samples - removed_samples);
removed_samples += add_samples;
afq->remaining_samples -= add_samples;
}
if (removed_samples > nb_samples)
av_log(afq->avctx, AV_LOG_WARNING, "frame_size is too large\n");
if (duration)
*duration = ff_samples_to_time_base(afq->avctx, removed_samples);
}
void ff_af_queue_log_state(AudioFrameQueue *afq)
{
AudioFrame *f;
av_log(afq->avctx, AV_LOG_DEBUG, "remaining delay = %d\n",
afq->remaining_delay);
av_log(afq->avctx, AV_LOG_DEBUG, "remaining samples = %d\n",
afq->remaining_samples);
av_log(afq->avctx, AV_LOG_DEBUG, "frames:\n");
f = afq->frame_queue;
while (f) {
av_log(afq->avctx, AV_LOG_DEBUG, " [ pts=%9"PRId64" duration=%d ]\n",
f->pts, f->duration);
f = f->next;
}
}
|