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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>
		
			
				
	
	
		
			220 lines
		
	
	
		
			5.9 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			220 lines
		
	
	
		
			5.9 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * AU muxer and demuxer
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|  * Copyright (c) 2001 Fabrice Bellard
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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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| 
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| /*
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|  * First version by Francois Revol revol@free.fr
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|  *
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|  * Reference documents:
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|  * http://www.opengroup.org/public/pubs/external/auformat.html
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|  * http://www.goice.co.jp/member/mo/formats/au.html
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|  */
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| 
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| #include "avformat.h"
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| #include "internal.h"
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| #include "avio_internal.h"
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| #include "pcm.h"
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| #include "riff.h"
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| 
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| /* if we don't know the size in advance */
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| #define AU_UNKNOWN_SIZE ((uint32_t)(~0))
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| 
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| /* The libavcodec codecs we support, and the IDs they have in the file */
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| static const AVCodecTag codec_au_tags[] = {
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|     { CODEC_ID_PCM_MULAW, 1 },
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|     { CODEC_ID_PCM_S8, 2 },
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|     { CODEC_ID_PCM_S16BE, 3 },
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|     { CODEC_ID_PCM_S24BE, 4 },
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|     { CODEC_ID_PCM_S32BE, 5 },
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|     { CODEC_ID_PCM_F32BE, 6 },
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|     { CODEC_ID_PCM_F64BE, 7 },
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|     { CODEC_ID_PCM_ALAW, 27 },
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|     { CODEC_ID_NONE, 0 },
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| };
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| 
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| #if CONFIG_AU_MUXER
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| /* AUDIO_FILE header */
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| static int put_au_header(AVIOContext *pb, AVCodecContext *enc)
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| {
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|     if(!enc->codec_tag)
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|         return -1;
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|     ffio_wfourcc(pb, ".snd");    /* magic number */
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|     avio_wb32(pb, 24);           /* header size */
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|     avio_wb32(pb, AU_UNKNOWN_SIZE); /* data size */
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|     avio_wb32(pb, (uint32_t)enc->codec_tag);     /* codec ID */
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|     avio_wb32(pb, enc->sample_rate);
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|     avio_wb32(pb, (uint32_t)enc->channels);
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|     return 0;
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| }
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| 
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| static int au_write_header(AVFormatContext *s)
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| {
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|     AVIOContext *pb = s->pb;
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| 
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|     s->priv_data = NULL;
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| 
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|     /* format header */
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|     if (put_au_header(pb, s->streams[0]->codec) < 0) {
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|         return -1;
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|     }
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| 
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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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| 
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| static int au_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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| 
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| static int au_write_trailer(AVFormatContext *s)
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| {
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|     AVIOContext *pb = s->pb;
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|     int64_t file_size;
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| 
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|     if (s->pb->seekable) {
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| 
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|         /* update file size */
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|         file_size = avio_tell(pb);
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|         avio_seek(pb, 8, SEEK_SET);
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|         avio_wb32(pb, (uint32_t)(file_size - 24));
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|         avio_seek(pb, file_size, SEEK_SET);
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| 
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|         avio_flush(pb);
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|     }
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| 
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|     return 0;
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| }
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| #endif /* CONFIG_AU_MUXER */
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| 
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| static int au_probe(AVProbeData *p)
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| {
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|     /* check file header */
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|     if (p->buf[0] == '.' && p->buf[1] == 's' &&
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|         p->buf[2] == 'n' && p->buf[3] == 'd')
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|         return AVPROBE_SCORE_MAX;
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|     else
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|         return 0;
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| }
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| 
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| /* au input */
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| static int au_read_header(AVFormatContext *s,
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|                           AVFormatParameters *ap)
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| {
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|     int size, bps, data_size = 0;
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|     unsigned int tag;
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|     AVIOContext *pb = s->pb;
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|     unsigned int id, channels, rate;
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|     enum CodecID codec;
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|     AVStream *st;
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| 
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|     /* check ".snd" header */
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|     tag = avio_rl32(pb);
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|     if (tag != MKTAG('.', 's', 'n', 'd'))
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|         return -1;
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|     size = avio_rb32(pb); /* header size */
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|     data_size = avio_rb32(pb); /* data size in bytes */
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| 
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|     if (data_size < 0 && data_size != AU_UNKNOWN_SIZE) {
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|         av_log(s, AV_LOG_ERROR, "Invalid negative data size '%d' found\n", data_size);
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     id = avio_rb32(pb);
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|     rate = avio_rb32(pb);
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|     channels = avio_rb32(pb);
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| 
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|     codec = ff_codec_get_id(codec_au_tags, id);
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| 
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|     if (!(bps = av_get_bits_per_sample(codec))) {
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|         av_log_ask_for_sample(s, "could not determine bits per sample\n");
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     if (size >= 24) {
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|         /* skip unused data */
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|         avio_skip(pb, size - 24);
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|     }
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| 
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|     /* now we are ready: build format streams */
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|     st = avformat_new_stream(s, NULL);
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|     if (!st)
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|         return -1;
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|     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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|     st->codec->codec_tag = id;
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|     st->codec->codec_id = codec;
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|     st->codec->channels = channels;
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|     st->codec->sample_rate = rate;
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|     if (data_size != AU_UNKNOWN_SIZE)
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|     st->duration = (((int64_t)data_size)<<3) / (st->codec->channels * bps);
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|     avpriv_set_pts_info(st, 64, 1, rate);
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|     return 0;
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| }
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| 
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| #define BLOCK_SIZE 1024
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| 
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| static int au_read_packet(AVFormatContext *s,
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|                           AVPacket *pkt)
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| {
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|     int ret;
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| 
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|     ret= av_get_packet(s->pb, pkt, BLOCK_SIZE *
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|                        s->streams[0]->codec->channels *
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|                        av_get_bits_per_sample(s->streams[0]->codec->codec_id) >> 3);
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|     if (ret < 0)
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|         return ret;
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|     pkt->stream_index = 0;
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| 
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|     /* note: we need to modify the packet size here to handle the last
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|        packet */
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|     pkt->size = ret;
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|     return 0;
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| }
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| 
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| #if CONFIG_AU_DEMUXER
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| AVInputFormat ff_au_demuxer = {
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|     .name           = "au",
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|     .long_name      = NULL_IF_CONFIG_SMALL("SUN AU format"),
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|     .read_probe     = au_probe,
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|     .read_header    = au_read_header,
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|     .read_packet    = au_read_packet,
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|     .read_seek      = pcm_read_seek,
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|     .codec_tag= (const AVCodecTag* const []){codec_au_tags, 0},
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| };
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| #endif
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| 
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| #if CONFIG_AU_MUXER
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| AVOutputFormat ff_au_muxer = {
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|     .name              = "au",
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|     .long_name         = NULL_IF_CONFIG_SMALL("SUN AU format"),
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|     .mime_type         = "audio/basic",
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|     .extensions        = "au",
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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      = au_write_header,
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|     .write_packet      = au_write_packet,
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|     .write_trailer     = au_write_trailer,
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|     .codec_tag= (const AVCodecTag* const []){codec_au_tags, 0},
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| };
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| #endif //CONFIG_AU_MUXER
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