annotate audiointerleave.c @ 4895:0fe3406b958a libavformat

Revert previous removal of gsize variable. It breaks seeking.
author diego
date Sun, 19 Apr 2009 11:40:03 +0000
parents fc0a165de804
children 536e5527c1e0
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1 /*
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2 * Audio Interleaving functions
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3 *
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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4 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
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5 *
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6 * This file is part of FFmpeg.
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7 *
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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8 * FFmpeg is free software; you can redistribute it and/or
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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9 * modify it under the terms of the GNU Lesser General Public
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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10 * License as published by the Free Software Foundation; either
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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11 * version 2.1 of the License, or (at your option) any later version.
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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12 *
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13 * FFmpeg is distributed in the hope that it will be useful,
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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16 * Lesser General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU Lesser General Public
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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19 * License along with FFmpeg; if not, write to the Free Software
65adb9e5214f extract audio interleaving code from mxf muxer, will be used by gxf and dv
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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21 */
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22
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23 #include "libavutil/fifo.h"
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24 #include "avformat.h"
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25 #include "audiointerleave.h"
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d5119d75439d Move declaration of ff_interleave_add_packet to internal.h.
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26 #include "internal.h"
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27
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28 void ff_audio_interleave_close(AVFormatContext *s)
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29 {
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30 int i;
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31 for (i = 0; i < s->nb_streams; i++) {
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32 AVStream *st = s->streams[i];
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33 AudioInterleaveContext *aic = st->priv_data;
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34
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35 if (st->codec->codec_type == CODEC_TYPE_AUDIO)
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36 av_fifo_free(aic->fifo);
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37 }
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38 }
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39
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40 int ff_audio_interleave_init(AVFormatContext *s,
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41 const int *samples_per_frame,
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42 AVRational time_base)
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43 {
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44 int i;
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45
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46 if (!samples_per_frame)
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47 return -1;
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48
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49 for (i = 0; i < s->nb_streams; i++) {
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50 AVStream *st = s->streams[i];
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51 AudioInterleaveContext *aic = st->priv_data;
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52
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53 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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54 aic->sample_size = (st->codec->channels *
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55 av_get_bits_per_sample(st->codec->codec_id)) / 8;
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56 if (!aic->sample_size) {
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57 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
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58 return -1;
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59 }
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60 aic->samples_per_frame = samples_per_frame;
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61 aic->samples = aic->samples_per_frame;
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62 aic->time_base = time_base;
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63
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64 aic->fifo_size = 100* *aic->samples;
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65 aic->fifo= av_fifo_alloc(100 * *aic->samples);
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66 }
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67 }
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68
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69 return 0;
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70 }
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71
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de5ccc7a4461 Declare ff_interleave_new_audio_packet as static, it is not used
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72 static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
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73 int stream_index, int flush)
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74 {
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75 AVStream *st = s->streams[stream_index];
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76 AudioInterleaveContext *aic = st->priv_data;
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77
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78 int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
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79 if (!size || (!flush && size == av_fifo_size(aic->fifo)))
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80 return 0;
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81
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82 av_new_packet(pkt, size);
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fc0a165de804 Reorder arguments for av_fifo_generic_read to be more logical and
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83 av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
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84
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85 pkt->dts = pkt->pts = aic->dts;
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86 pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
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87 pkt->stream_index = stream_index;
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88 aic->dts += pkt->duration;
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89
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90 aic->samples++;
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91 if (!*aic->samples)
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92 aic->samples = aic->samples_per_frame;
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93
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94 return size;
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95 }
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96
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97 int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
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98 int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
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99 int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
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100 {
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101 int i;
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102
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103 if (pkt) {
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104 AVStream *st = s->streams[pkt->stream_index];
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105 AudioInterleaveContext *aic = st->priv_data;
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106 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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107 unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
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108 if (new_size > aic->fifo_size) {
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109 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
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110 return -1;
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111 aic->fifo_size = new_size;
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112 }
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113 av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
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114 } else {
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115 // rewrite pts and dts to be decoded time line position
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116 pkt->pts = pkt->dts = aic->dts;
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117 aic->dts += pkt->duration;
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118 ff_interleave_add_packet(s, pkt, compare_ts);
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119 }
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120 pkt = NULL;
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121 }
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122
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123 for (i = 0; i < s->nb_streams; i++) {
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124 AVStream *st = s->streams[i];
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125 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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126 AVPacket new_pkt;
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127 while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
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128 ff_interleave_add_packet(s, &new_pkt, compare_ts);
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129 }
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130 }
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131
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132 return get_packet(s, out, pkt, flush);
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133 }