| 1 | /* |
| 2 | * Audio Interleaving functions |
| 3 | * |
| 4 | * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> |
| 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_freep(&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 AVERROR(EINVAL); |
| 49 | |
| 50 | if (!time_base.num) { |
| 51 | av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); |
| 52 | return AVERROR(EINVAL); |
| 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 AVERROR(EINVAL); |
| 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 | if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples))) |
| 71 | return AVERROR(ENOMEM); |
| 72 | } |
| 73 | } |
| 74 | |
| 75 | return 0; |
| 76 | } |
| 77 | |
| 78 | static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, |
| 79 | int stream_index, int flush) |
| 80 | { |
| 81 | AVStream *st = s->streams[stream_index]; |
| 82 | AudioInterleaveContext *aic = st->priv_data; |
| 83 | int ret; |
| 84 | int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size); |
| 85 | if (!size || (!flush && size == av_fifo_size(aic->fifo))) |
| 86 | return 0; |
| 87 | |
| 88 | ret = av_new_packet(pkt, size); |
| 89 | if (ret < 0) |
| 90 | return ret; |
| 91 | av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); |
| 92 | |
| 93 | pkt->dts = pkt->pts = aic->dts; |
| 94 | pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); |
| 95 | pkt->stream_index = stream_index; |
| 96 | aic->dts += pkt->duration; |
| 97 | |
| 98 | aic->samples++; |
| 99 | if (!*aic->samples) |
| 100 | aic->samples = aic->samples_per_frame; |
| 101 | |
| 102 | return size; |
| 103 | } |
| 104 | |
| 105 | int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, |
| 106 | int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), |
| 107 | int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)) |
| 108 | { |
| 109 | int i, ret; |
| 110 | |
| 111 | if (pkt) { |
| 112 | AVStream *st = s->streams[pkt->stream_index]; |
| 113 | AudioInterleaveContext *aic = st->priv_data; |
| 114 | if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
| 115 | unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; |
| 116 | if (new_size > aic->fifo_size) { |
| 117 | if (av_fifo_realloc2(aic->fifo, new_size) < 0) |
| 118 | return AVERROR(ENOMEM); |
| 119 | aic->fifo_size = new_size; |
| 120 | } |
| 121 | av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); |
| 122 | } else { |
| 123 | // rewrite pts and dts to be decoded time line position |
| 124 | pkt->pts = pkt->dts = aic->dts; |
| 125 | aic->dts += pkt->duration; |
| 126 | if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 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 | while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { |
| 137 | if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0) |
| 138 | return ret; |
| 139 | } |
| 140 | if (ret < 0) |
| 141 | return ret; |
| 142 | } |
| 143 | } |
| 144 | |
| 145 | return get_packet(s, out, NULL, flush); |
| 146 | } |