FFmpeg: libavformat/audiointerleave.c Source File

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libavformat

audiointerleave.c

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1 /*

2 * Audio Interleaving functions

3 *

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 */

23 #include "libavutil/fifo.h"

24 #include "libavutil/mathematics.h"

25 #include "avformat.h"

26 #include "audiointerleave.h"

27 #include "internal.h"

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;

36 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)

37 av_fifo_free(aic->fifo);

38 }

39 }

41 int ff_audio_interleave_init(AVFormatContext *s,

42 const int *samples_per_frame,

43 AVRational time_base)

44 {

45 int i;

47 if (!samples_per_frame)

48 return -1;

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;

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;

69 aic->fifo_size = 100* *aic->samples;

70 aic->fifo= av_fifo_alloc(100 * *aic->samples);

71 }

72 }

74 return 0;

75 }

77 static int ff_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;

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;

87 if (av_new_packet(pkt, size) < 0)

88 return AVERROR(ENOMEM);

89 av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);

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;

96 aic->samples++;

97 if (!*aic->samples)

98 aic->samples = aic->samples_per_frame;

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 = ff_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 }

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