Subversion Repositories Kolibri OS

Rev

Details | Last modification | View Log | RSS feed

Rev Author Line No. Line
6148 serge 1 
/*
2 
 * Audio Interleaving functions
3 
 *
4 
 * Copyright (c) 2009 Baptiste Coudurier
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 
 
29 
void ff_audio_interleave_close(AVFormatContext *s)
30 
{
31 
    int i;
32 
    for (i = 0; i < s->nb_streams; i++) {
33 
        AVStream *st = s->streams[i];
34 
        AudioInterleaveContext *aic = st->priv_data;
35 
 
36 
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
37 
            av_fifo_free(aic->fifo);
38 
    }
39 
}
40 
 
41 
int ff_audio_interleave_init(AVFormatContext *s,
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 
    if (!time_base.num) {
51 
        av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
52 
        return -1;
53 
    }
54 
    for (i = 0; i < s->nb_streams; i++) {
55 
        AVStream *st = s->streams[i];
56 
        AudioInterleaveContext *aic = st->priv_data;
57 
 
58 
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
59 
            aic->sample_size = (st->codec->channels *
60 
                                av_get_bits_per_sample(st->codec->codec_id)) / 8;
61 
            if (!aic->sample_size) {
62 
                av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
63 
                return -1;
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 
            aic->fifo= av_fifo_alloc(100 * *aic->samples);
71 
        }
72 
    }
73 
 
74 
    return 0;
75 
}
76 
 
77 
static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
78 
                                       int stream_index, int flush)
79 
{
80 
    AVStream *st = s->streams[stream_index];
81 
    AudioInterleaveContext *aic = st->priv_data;
82 
 
83 
    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
84 
    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
85 
        return 0;
86 
 
87 
    if (av_new_packet(pkt, size) < 0)
88 
        return AVERROR(ENOMEM);
89 
    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
90 
 
91 
    pkt->dts = pkt->pts = aic->dts;
92 
    pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
93 
    pkt->stream_index = stream_index;
94 
    aic->dts += pkt->duration;
95 
 
96 
    aic->samples++;
97 
    if (!*aic->samples)
98 
        aic->samples = aic->samples_per_frame;
99 
 
100 
    return size;
101 
}
102 
 
103 
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
104 
                        int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
105 
                        int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
106 
{
107 
    int i;
108 
 
109 
    if (pkt) {
110 
        AVStream *st = s->streams[pkt->stream_index];
111 
        AudioInterleaveContext *aic = st->priv_data;
112 
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
113 
            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
114 
            if (new_size > aic->fifo_size) {
115 
                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
116 
                    return -1;
117 
                aic->fifo_size = new_size;
118 
            }
119 
            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
120 
        } else {
121 
            int ret;
122 
            // rewrite pts and dts to be decoded time line position
123 
            pkt->pts = pkt->dts = aic->dts;
124 
            aic->dts += pkt->duration;
125 
            ret = ff_interleave_add_packet(s, pkt, compare_ts);
126 
            if (ret < 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->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
135 
            AVPacket new_pkt;
136 
            int ret;
137 
            while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
138 
                ret = ff_interleave_add_packet(s, &new_pkt, compare_ts);
139 
                if (ret < 0)
140 
                    return ret;
141 
            }
142 
            if (ret < 0)
143 
                return ret;
144 
        }
145 
    }
146 
 
147 
    return get_packet(s, out, NULL, flush);
148 
}