Mercurial > libavformat.hg
annotate audiointerleave.c @ 5079:cc9c5e6b8ccf libavformat
Use av_log_missing_feature in the ADTS encoder. (Pseudo reverts r19173).
author | alexc |
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date | Mon, 29 Jun 2009 22:48:07 +0000 |
parents | fc0a165de804 |
children | 536e5527c1e0 |
rev | line source |
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1 /* |
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2 * Audio Interleaving functions |
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3 * |
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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 * |
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8 * FFmpeg is free software; you can redistribute it and/or |
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9 * modify it under the terms of the GNU Lesser General Public |
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10 * License as published by the Free Software Foundation; either |
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11 * version 2.1 of the License, or (at your option) any later version. |
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12 * |
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13 * FFmpeg is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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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 |
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19 * License along with FFmpeg; if not, write to the Free Software |
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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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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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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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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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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; |
4534 | 108 if (new_size > aic->fifo_size) { |
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109 if (av_fifo_realloc2(aic->fifo, new_size) < 0) |
4534 | 110 return -1; |
111 aic->fifo_size = new_size; | |
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 |
4409 | 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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extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
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127 while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) |
65adb9e5214f
extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
diff
changeset
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128 ff_interleave_add_packet(s, &new_pkt, compare_ts); |
65adb9e5214f
extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
diff
changeset
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129 } |
65adb9e5214f
extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
diff
changeset
|
130 } |
65adb9e5214f
extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
diff
changeset
|
131 |
65adb9e5214f
extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
diff
changeset
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132 return get_packet(s, out, pkt, flush); |
65adb9e5214f
extract audio interleaving code from mxf muxer, will be used by gxf and dv
bcoudurier
parents:
diff
changeset
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133 } |