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9d76cf0b18
* qatar/master: rtpdec: Templatize the code for different g726 bitrate variants rv40: move loop filter to rv34dsp context lavf: make av_set_pts_info private. rtpdec: Add support for G726 audio rtpdec: Add an init function that can do custom codec context initialization avconv: make copy_tb on by default. matroskadec: don't set codec timebase. rmdec: don't set codec timebase. avconv: compute next_pts from input packet duration when possible. lavf: estimate frame duration from r_frame_rate. avconv: update InputStream.pts in the streamcopy case. Conflicts: avconv.c libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/oss_audio.c libavdevice/v4l.c libavdevice/v4l2.c libavdevice/vfwcap.c libavdevice/x11grab.c libavformat/au.c libavformat/eacdata.c libavformat/flvdec.c libavformat/mpegts.c libavformat/mxfenc.c libavformat/rtpdec_g726.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
171 lines
5.0 KiB
C
171 lines
5.0 KiB
C
/*
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* AIFF/AIFF-C muxer
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* Copyright (c) 2006 Patrick Guimond
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intfloat_readwrite.h"
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#include "avformat.h"
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#include "internal.h"
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#include "aiff.h"
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#include "avio_internal.h"
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#include "isom.h"
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typedef struct {
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int64_t form;
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int64_t frames;
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int64_t ssnd;
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} AIFFOutputContext;
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static int aiff_write_header(AVFormatContext *s)
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{
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AIFFOutputContext *aiff = s->priv_data;
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AVIOContext *pb = s->pb;
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AVCodecContext *enc = s->streams[0]->codec;
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AVExtFloat sample_rate;
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int aifc = 0;
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/* First verify if format is ok */
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if (!enc->codec_tag)
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return -1;
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if (enc->codec_tag != MKTAG('N','O','N','E'))
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aifc = 1;
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/* FORM AIFF header */
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ffio_wfourcc(pb, "FORM");
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aiff->form = avio_tell(pb);
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avio_wb32(pb, 0); /* file length */
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ffio_wfourcc(pb, aifc ? "AIFC" : "AIFF");
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if (aifc) { // compressed audio
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enc->bits_per_coded_sample = 16;
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if (!enc->block_align) {
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av_log(s, AV_LOG_ERROR, "block align not set\n");
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return -1;
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}
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/* Version chunk */
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ffio_wfourcc(pb, "FVER");
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avio_wb32(pb, 4);
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avio_wb32(pb, 0xA2805140);
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}
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if (enc->channels > 2 && enc->channel_layout) {
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ffio_wfourcc(pb, "CHAN");
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avio_wb32(pb, 12);
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ff_mov_write_chan(pb, enc->channel_layout);
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}
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/* Common chunk */
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ffio_wfourcc(pb, "COMM");
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avio_wb32(pb, aifc ? 24 : 18); /* size */
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avio_wb16(pb, enc->channels); /* Number of channels */
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aiff->frames = avio_tell(pb);
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avio_wb32(pb, 0); /* Number of frames */
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if (!enc->bits_per_coded_sample)
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enc->bits_per_coded_sample = av_get_bits_per_sample(enc->codec_id);
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if (!enc->bits_per_coded_sample) {
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av_log(s, AV_LOG_ERROR, "could not compute bits per sample\n");
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return -1;
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}
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if (!enc->block_align)
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enc->block_align = (enc->bits_per_coded_sample * enc->channels) >> 3;
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avio_wb16(pb, enc->bits_per_coded_sample); /* Sample size */
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sample_rate = av_dbl2ext((double)enc->sample_rate);
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avio_write(pb, (uint8_t*)&sample_rate, sizeof(sample_rate));
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if (aifc) {
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avio_wl32(pb, enc->codec_tag);
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avio_wb16(pb, 0);
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}
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/* Sound data chunk */
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ffio_wfourcc(pb, "SSND");
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aiff->ssnd = avio_tell(pb); /* Sound chunk size */
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avio_wb32(pb, 0); /* Sound samples data size */
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avio_wb32(pb, 0); /* Data offset */
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avio_wb32(pb, 0); /* Block-size (block align) */
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avpriv_set_pts_info(s->streams[0], 64, 1, s->streams[0]->codec->sample_rate);
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/* Data is starting here */
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avio_flush(pb);
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return 0;
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}
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static int aiff_write_packet(AVFormatContext *s, AVPacket *pkt)
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{
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AVIOContext *pb = s->pb;
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avio_write(pb, pkt->data, pkt->size);
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return 0;
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}
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static int aiff_write_trailer(AVFormatContext *s)
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{
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AVIOContext *pb = s->pb;
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AIFFOutputContext *aiff = s->priv_data;
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AVCodecContext *enc = s->streams[0]->codec;
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/* Chunks sizes must be even */
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int64_t file_size, end_size;
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end_size = file_size = avio_tell(pb);
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if (file_size & 1) {
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avio_w8(pb, 0);
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end_size++;
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}
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if (s->pb->seekable) {
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/* File length */
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avio_seek(pb, aiff->form, SEEK_SET);
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avio_wb32(pb, file_size - aiff->form - 4);
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/* Number of sample frames */
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avio_seek(pb, aiff->frames, SEEK_SET);
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avio_wb32(pb, (file_size-aiff->ssnd-12)/enc->block_align);
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/* Sound Data chunk size */
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avio_seek(pb, aiff->ssnd, SEEK_SET);
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avio_wb32(pb, file_size - aiff->ssnd - 4);
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/* return to the end */
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avio_seek(pb, end_size, SEEK_SET);
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avio_flush(pb);
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}
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return 0;
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}
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AVOutputFormat ff_aiff_muxer = {
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.name = "aiff",
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.long_name = NULL_IF_CONFIG_SMALL("Audio IFF"),
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.mime_type = "audio/aiff",
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.extensions = "aif,aiff,afc,aifc",
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.priv_data_size = sizeof(AIFFOutputContext),
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.audio_codec = CODEC_ID_PCM_S16BE,
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.video_codec = CODEC_ID_NONE,
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.write_header = aiff_write_header,
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.write_packet = aiff_write_packet,
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.write_trailer = aiff_write_trailer,
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.codec_tag= (const AVCodecTag* const []){ff_codec_aiff_tags, 0},
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};
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