1075 lines
38 KiB
C
1075 lines
38 KiB
C
/*
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* RTP input/output format
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* Copyright (c) 2002 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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#include "avformat.h"
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#include "mpegts.h"
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#include "bitstream.h"
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#include <unistd.h>
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#include "network.h"
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#include "rtp_internal.h"
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#include "rtp_h264.h"
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#include "rtp_mpv.h"
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#include "rtp_aac.h"
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//#define DEBUG
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#define RTCP_SR_SIZE 28
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/* TODO: - add RTCP statistics reporting (should be optional).
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- add support for h263/mpeg4 packetized output : IDEA: send a
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buffer to 'rtp_write_packet' contains all the packets for ONE
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frame. Each packet should have a four byte header containing
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the length in big endian format (same trick as
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'url_open_dyn_packet_buf')
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*/
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/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
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AVRtpPayloadType_t AVRtpPayloadTypes[]=
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{
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{0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1},
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{1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
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{7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1},
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{9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
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{11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
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{12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},
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{13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
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{15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1},
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{17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1},
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{18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
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{19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
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{20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
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{21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
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{22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
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{23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
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{24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
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{25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
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{26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1},
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{27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
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{28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
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{29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
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{30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
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{31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1},
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{32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1},
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{32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG2VIDEO, 90000, -1},
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{33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1},
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{34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1},
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{35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
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{-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
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};
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/* statistics functions */
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RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
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static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
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static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
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static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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{
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handler->next= RTPFirstDynamicPayloadHandler;
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RTPFirstDynamicPayloadHandler= handler;
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}
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void av_register_rtp_dynamic_payload_handlers(void)
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{
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register_dynamic_payload_handler(&mp4v_es_handler);
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register_dynamic_payload_handler(&mpeg4_generic_handler);
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register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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}
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int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
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{
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int i = 0;
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for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
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if (AVRtpPayloadTypes[i].pt == payload_type) {
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if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) {
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codec->codec_type = AVRtpPayloadTypes[i].codec_type;
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codec->codec_id = AVRtpPayloadTypes[i].codec_id;
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if (AVRtpPayloadTypes[i].audio_channels > 0)
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codec->channels = AVRtpPayloadTypes[i].audio_channels;
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if (AVRtpPayloadTypes[i].clock_rate > 0)
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codec->sample_rate = AVRtpPayloadTypes[i].clock_rate;
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return 0;
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}
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}
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return -1;
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}
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int rtp_get_payload_type(AVCodecContext *codec)
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{
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int i, payload_type;
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/* compute the payload type */
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for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
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if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
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if (codec->codec_id == CODEC_ID_PCM_S16BE)
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if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
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continue;
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payload_type = AVRtpPayloadTypes[i].pt;
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}
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return payload_type;
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}
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const char *ff_rtp_enc_name(int payload_type)
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{
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int i;
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for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
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if (AVRtpPayloadTypes[i].pt == payload_type) {
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return AVRtpPayloadTypes[i].enc_name;
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}
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return "";
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}
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enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
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{
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int i;
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for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
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if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){
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return AVRtpPayloadTypes[i].codec_id;
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}
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return CODEC_ID_NONE;
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}
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
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{
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if (buf[1] != 200)
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return -1;
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s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
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s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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s->last_rtcp_timestamp = AV_RB32(buf + 16);
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return 0;
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}
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#define RTP_SEQ_MOD (1<<16)
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/**
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* called on parse open packet
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*/
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static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
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{
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memset(s, 0, sizeof(RTPStatistics));
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s->max_seq= base_sequence;
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s->probation= 1;
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}
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/**
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* called whenever there is a large jump in sequence numbers, or when they get out of probation...
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*/
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static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
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{
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s->max_seq= seq;
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s->cycles= 0;
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s->base_seq= seq -1;
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s->bad_seq= RTP_SEQ_MOD + 1;
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s->received= 0;
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s->expected_prior= 0;
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s->received_prior= 0;
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s->jitter= 0;
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s->transit= 0;
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}
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/**
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* returns 1 if we should handle this packet.
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*/
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static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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{
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uint16_t udelta= seq - s->max_seq;
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const int MAX_DROPOUT= 3000;
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const int MAX_MISORDER = 100;
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const int MIN_SEQUENTIAL = 2;
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/* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
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if(s->probation)
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{
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if(seq==s->max_seq + 1) {
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s->probation--;
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s->max_seq= seq;
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if(s->probation==0) {
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rtp_init_sequence(s, seq);
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s->received++;
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return 1;
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}
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} else {
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s->probation= MIN_SEQUENTIAL - 1;
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s->max_seq = seq;
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}
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} else if (udelta < MAX_DROPOUT) {
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// in order, with permissible gap
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if(seq < s->max_seq) {
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//sequence number wrapped; count antother 64k cycles
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s->cycles += RTP_SEQ_MOD;
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}
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s->max_seq= seq;
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} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
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// sequence made a large jump...
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if(seq==s->bad_seq) {
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// two sequential packets-- assume that the other side restarted without telling us; just resync.
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rtp_init_sequence(s, seq);
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} else {
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s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
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return 0;
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}
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} else {
|
|
// duplicate or reordered packet...
|
|
}
|
|
s->received++;
|
|
return 1;
|
|
}
|
|
|
|
#if 0
|
|
/**
|
|
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
|
|
* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
|
|
* never change. I left this in in case someone else can see a way. (rdm)
|
|
*/
|
|
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
|
|
{
|
|
uint32_t transit= arrival_timestamp - sent_timestamp;
|
|
int d;
|
|
s->transit= transit;
|
|
d= FFABS(transit - s->transit);
|
|
s->jitter += d - ((s->jitter + 8)>>4);
|
|
}
|
|
#endif
|
|
|
|
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
|
|
{
|
|
ByteIOContext pb;
|
|
uint8_t *buf;
|
|
int len;
|
|
int rtcp_bytes;
|
|
RTPStatistics *stats= &s->statistics;
|
|
uint32_t lost;
|
|
uint32_t extended_max;
|
|
uint32_t expected_interval;
|
|
uint32_t received_interval;
|
|
uint32_t lost_interval;
|
|
uint32_t expected;
|
|
uint32_t fraction;
|
|
uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
|
|
|
|
if (!s->rtp_ctx || (count < 1))
|
|
return -1;
|
|
|
|
/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
|
|
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
|
|
s->octet_count += count;
|
|
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
|
|
RTCP_TX_RATIO_DEN;
|
|
rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
|
|
if (rtcp_bytes < 28)
|
|
return -1;
|
|
s->last_octet_count = s->octet_count;
|
|
|
|
if (url_open_dyn_buf(&pb) < 0)
|
|
return -1;
|
|
|
|
// Receiver Report
|
|
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
|
|
put_byte(&pb, 201);
|
|
put_be16(&pb, 7); /* length in words - 1 */
|
|
put_be32(&pb, s->ssrc); // our own SSRC
|
|
put_be32(&pb, s->ssrc); // XXX: should be the server's here!
|
|
// some placeholders we should really fill...
|
|
// RFC 1889/p64
|
|
extended_max= stats->cycles + stats->max_seq;
|
|
expected= extended_max - stats->base_seq + 1;
|
|
lost= expected - stats->received;
|
|
lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
|
|
expected_interval= expected - stats->expected_prior;
|
|
stats->expected_prior= expected;
|
|
received_interval= stats->received - stats->received_prior;
|
|
stats->received_prior= stats->received;
|
|
lost_interval= expected_interval - received_interval;
|
|
if (expected_interval==0 || lost_interval<=0) fraction= 0;
|
|
else fraction = (lost_interval<<8)/expected_interval;
|
|
|
|
fraction= (fraction<<24) | lost;
|
|
|
|
put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
|
|
put_be32(&pb, extended_max); /* max sequence received */
|
|
put_be32(&pb, stats->jitter>>4); /* jitter */
|
|
|
|
if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
|
|
{
|
|
put_be32(&pb, 0); /* last SR timestamp */
|
|
put_be32(&pb, 0); /* delay since last SR */
|
|
} else {
|
|
uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
|
|
uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
|
|
|
|
put_be32(&pb, middle_32_bits); /* last SR timestamp */
|
|
put_be32(&pb, delay_since_last); /* delay since last SR */
|
|
}
|
|
|
|
// CNAME
|
|
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
|
|
put_byte(&pb, 202);
|
|
len = strlen(s->hostname);
|
|
put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
|
|
put_be32(&pb, s->ssrc);
|
|
put_byte(&pb, 0x01);
|
|
put_byte(&pb, len);
|
|
put_buffer(&pb, s->hostname, len);
|
|
// padding
|
|
for (len = (6 + len) % 4; len % 4; len++) {
|
|
put_byte(&pb, 0);
|
|
}
|
|
|
|
put_flush_packet(&pb);
|
|
len = url_close_dyn_buf(&pb, &buf);
|
|
if ((len > 0) && buf) {
|
|
int result;
|
|
#if defined(DEBUG)
|
|
printf("sending %d bytes of RR\n", len);
|
|
#endif
|
|
result= url_write(s->rtp_ctx, buf, len);
|
|
#if defined(DEBUG)
|
|
printf("result from url_write: %d\n", result);
|
|
#endif
|
|
av_free(buf);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
|
|
* MPEG2TS streams to indicate that they should be demuxed inside the
|
|
* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
|
|
* TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
|
|
*/
|
|
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
|
|
{
|
|
RTPDemuxContext *s;
|
|
|
|
s = av_mallocz(sizeof(RTPDemuxContext));
|
|
if (!s)
|
|
return NULL;
|
|
s->payload_type = payload_type;
|
|
s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
s->ic = s1;
|
|
s->st = st;
|
|
s->rtp_payload_data = rtp_payload_data;
|
|
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
|
|
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
|
|
s->ts = mpegts_parse_open(s->ic);
|
|
if (s->ts == NULL) {
|
|
av_free(s);
|
|
return NULL;
|
|
}
|
|
} else {
|
|
switch(st->codec->codec_id) {
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
case CODEC_ID_MP2:
|
|
case CODEC_ID_MP3:
|
|
case CODEC_ID_MPEG4:
|
|
case CODEC_ID_H264:
|
|
st->need_parsing = AVSTREAM_PARSE_FULL;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
// needed to send back RTCP RR in RTSP sessions
|
|
s->rtp_ctx = rtpc;
|
|
gethostname(s->hostname, sizeof(s->hostname));
|
|
return s;
|
|
}
|
|
|
|
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
|
|
{
|
|
int au_headers_length, au_header_size, i;
|
|
GetBitContext getbitcontext;
|
|
rtp_payload_data_t *infos;
|
|
|
|
infos = s->rtp_payload_data;
|
|
|
|
if (infos == NULL)
|
|
return -1;
|
|
|
|
/* decode the first 2 bytes where are stored the AUHeader sections
|
|
length in bits */
|
|
au_headers_length = AV_RB16(buf);
|
|
|
|
if (au_headers_length > RTP_MAX_PACKET_LENGTH)
|
|
return -1;
|
|
|
|
infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
|
|
|
|
/* skip AU headers length section (2 bytes) */
|
|
buf += 2;
|
|
|
|
init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
|
|
|
|
/* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
|
|
au_header_size = infos->sizelength + infos->indexlength;
|
|
if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
|
|
return -1;
|
|
|
|
infos->nb_au_headers = au_headers_length / au_header_size;
|
|
infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
|
|
|
|
/* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
|
|
In my test, the FAAD decoder does not behave correctly when sending each AU one by one
|
|
but does when sending the whole as one big packet... */
|
|
infos->au_headers[0].size = 0;
|
|
infos->au_headers[0].index = 0;
|
|
for (i = 0; i < infos->nb_au_headers; ++i) {
|
|
infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
|
|
infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
|
|
}
|
|
|
|
infos->nb_au_headers = 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
|
|
*/
|
|
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
|
|
{
|
|
switch(s->st->codec->codec_id) {
|
|
case CODEC_ID_MP2:
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
|
|
int64_t addend;
|
|
|
|
int delta_timestamp;
|
|
/* XXX: is it really necessary to unify the timestamp base ? */
|
|
/* compute pts from timestamp with received ntp_time */
|
|
delta_timestamp = timestamp - s->last_rtcp_timestamp;
|
|
/* convert to 90 kHz without overflow */
|
|
addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
|
|
addend = (addend * 5625) >> 14;
|
|
pkt->pts = addend + delta_timestamp;
|
|
}
|
|
break;
|
|
case CODEC_ID_AAC:
|
|
case CODEC_ID_H264:
|
|
case CODEC_ID_MPEG4:
|
|
pkt->pts = timestamp;
|
|
break;
|
|
default:
|
|
/* no timestamp info yet */
|
|
break;
|
|
}
|
|
pkt->stream_index = s->st->index;
|
|
}
|
|
|
|
/**
|
|
* Parse an RTP or RTCP packet directly sent as a buffer.
|
|
* @param s RTP parse context.
|
|
* @param pkt returned packet
|
|
* @param buf input buffer or NULL to read the next packets
|
|
* @param len buffer len
|
|
* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
|
|
* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
|
|
*/
|
|
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
|
|
const uint8_t *buf, int len)
|
|
{
|
|
unsigned int ssrc, h;
|
|
int payload_type, seq, ret;
|
|
AVStream *st;
|
|
uint32_t timestamp;
|
|
int rv= 0;
|
|
|
|
if (!buf) {
|
|
/* return the next packets, if any */
|
|
if(s->st && s->parse_packet) {
|
|
timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
|
|
rv= s->parse_packet(s, pkt, ×tamp, NULL, 0);
|
|
finalize_packet(s, pkt, timestamp);
|
|
return rv;
|
|
} else {
|
|
// TODO: Move to a dynamic packet handler (like above)
|
|
if (s->read_buf_index >= s->read_buf_size)
|
|
return -1;
|
|
ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
|
|
s->read_buf_size - s->read_buf_index);
|
|
if (ret < 0)
|
|
return -1;
|
|
s->read_buf_index += ret;
|
|
if (s->read_buf_index < s->read_buf_size)
|
|
return 1;
|
|
else
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (len < 12)
|
|
return -1;
|
|
|
|
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
|
|
return -1;
|
|
if (buf[1] >= 200 && buf[1] <= 204) {
|
|
rtcp_parse_packet(s, buf, len);
|
|
return -1;
|
|
}
|
|
payload_type = buf[1] & 0x7f;
|
|
seq = AV_RB16(buf + 2);
|
|
timestamp = AV_RB32(buf + 4);
|
|
ssrc = AV_RB32(buf + 8);
|
|
/* store the ssrc in the RTPDemuxContext */
|
|
s->ssrc = ssrc;
|
|
|
|
/* NOTE: we can handle only one payload type */
|
|
if (s->payload_type != payload_type)
|
|
return -1;
|
|
|
|
st = s->st;
|
|
// only do something with this if all the rtp checks pass...
|
|
if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
|
|
{
|
|
av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
|
|
payload_type, seq, ((s->seq + 1) & 0xffff));
|
|
return -1;
|
|
}
|
|
|
|
s->seq = seq;
|
|
len -= 12;
|
|
buf += 12;
|
|
|
|
if (!st) {
|
|
/* specific MPEG2TS demux support */
|
|
ret = mpegts_parse_packet(s->ts, pkt, buf, len);
|
|
if (ret < 0)
|
|
return -1;
|
|
if (ret < len) {
|
|
s->read_buf_size = len - ret;
|
|
memcpy(s->buf, buf + ret, s->read_buf_size);
|
|
s->read_buf_index = 0;
|
|
return 1;
|
|
}
|
|
} else {
|
|
// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
|
|
switch(st->codec->codec_id) {
|
|
case CODEC_ID_MP2:
|
|
/* better than nothing: skip mpeg audio RTP header */
|
|
if (len <= 4)
|
|
return -1;
|
|
h = AV_RB32(buf);
|
|
len -= 4;
|
|
buf += 4;
|
|
av_new_packet(pkt, len);
|
|
memcpy(pkt->data, buf, len);
|
|
break;
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
/* better than nothing: skip mpeg video RTP header */
|
|
if (len <= 4)
|
|
return -1;
|
|
h = AV_RB32(buf);
|
|
buf += 4;
|
|
len -= 4;
|
|
if (h & (1 << 26)) {
|
|
/* mpeg2 */
|
|
if (len <= 4)
|
|
return -1;
|
|
buf += 4;
|
|
len -= 4;
|
|
}
|
|
av_new_packet(pkt, len);
|
|
memcpy(pkt->data, buf, len);
|
|
break;
|
|
// moved from below, verbatim. this is because this section handles packets, and the lower switch handles
|
|
// timestamps.
|
|
// TODO: Put this into a dynamic packet handler...
|
|
case CODEC_ID_AAC:
|
|
if (rtp_parse_mp4_au(s, buf))
|
|
return -1;
|
|
{
|
|
rtp_payload_data_t *infos = s->rtp_payload_data;
|
|
if (infos == NULL)
|
|
return -1;
|
|
buf += infos->au_headers_length_bytes + 2;
|
|
len -= infos->au_headers_length_bytes + 2;
|
|
|
|
/* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
|
|
one au_header */
|
|
av_new_packet(pkt, infos->au_headers[0].size);
|
|
memcpy(pkt->data, buf, infos->au_headers[0].size);
|
|
buf += infos->au_headers[0].size;
|
|
len -= infos->au_headers[0].size;
|
|
}
|
|
s->read_buf_size = len;
|
|
rv= 0;
|
|
break;
|
|
default:
|
|
if(s->parse_packet) {
|
|
rv= s->parse_packet(s, pkt, ×tamp, buf, len);
|
|
} else {
|
|
av_new_packet(pkt, len);
|
|
memcpy(pkt->data, buf, len);
|
|
}
|
|
break;
|
|
}
|
|
|
|
// now perform timestamp things....
|
|
finalize_packet(s, pkt, timestamp);
|
|
}
|
|
return rv;
|
|
}
|
|
|
|
void rtp_parse_close(RTPDemuxContext *s)
|
|
{
|
|
// TODO: fold this into the protocol specific data fields.
|
|
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
|
|
mpegts_parse_close(s->ts);
|
|
}
|
|
av_free(s);
|
|
}
|
|
|
|
/* rtp output */
|
|
|
|
static int rtp_write_header(AVFormatContext *s1)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
int payload_type, max_packet_size, n;
|
|
AVStream *st;
|
|
|
|
if (s1->nb_streams != 1)
|
|
return -1;
|
|
st = s1->streams[0];
|
|
|
|
payload_type = rtp_get_payload_type(st->codec);
|
|
if (payload_type < 0)
|
|
payload_type = RTP_PT_PRIVATE; /* private payload type */
|
|
s->payload_type = payload_type;
|
|
|
|
// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
|
|
s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
|
|
s->timestamp = s->base_timestamp;
|
|
s->cur_timestamp = 0;
|
|
s->ssrc = 0; /* FIXME: was random(), what should this be? */
|
|
s->first_packet = 1;
|
|
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
|
|
max_packet_size = url_fget_max_packet_size(&s1->pb);
|
|
if (max_packet_size <= 12)
|
|
return AVERROR(EIO);
|
|
s->max_payload_size = max_packet_size - 12;
|
|
|
|
s->max_frames_per_packet = 0;
|
|
if (s1->max_delay) {
|
|
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
|
|
if (st->codec->frame_size == 0) {
|
|
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
|
|
} else {
|
|
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
|
|
}
|
|
}
|
|
if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
|
|
/* FIXME: We should round down here... */
|
|
s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
|
|
}
|
|
}
|
|
|
|
av_set_pts_info(st, 32, 1, 90000);
|
|
switch(st->codec->codec_id) {
|
|
case CODEC_ID_MP2:
|
|
case CODEC_ID_MP3:
|
|
s->buf_ptr = s->buf + 4;
|
|
break;
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
break;
|
|
case CODEC_ID_MPEG2TS:
|
|
n = s->max_payload_size / TS_PACKET_SIZE;
|
|
if (n < 1)
|
|
n = 1;
|
|
s->max_payload_size = n * TS_PACKET_SIZE;
|
|
s->buf_ptr = s->buf;
|
|
break;
|
|
case CODEC_ID_AAC:
|
|
s->read_buf_index = 0;
|
|
default:
|
|
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
|
|
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
|
|
}
|
|
s->buf_ptr = s->buf;
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* send an rtcp sender report packet */
|
|
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
uint32_t rtp_ts;
|
|
|
|
#if defined(DEBUG)
|
|
printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
|
|
#endif
|
|
|
|
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
|
|
s->last_rtcp_ntp_time = ntp_time;
|
|
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
|
|
s1->streams[0]->time_base) + s->base_timestamp;
|
|
put_byte(&s1->pb, (RTP_VERSION << 6));
|
|
put_byte(&s1->pb, 200);
|
|
put_be16(&s1->pb, 6); /* length in words - 1 */
|
|
put_be32(&s1->pb, s->ssrc);
|
|
put_be32(&s1->pb, ntp_time / 1000000);
|
|
put_be32(&s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
|
|
put_be32(&s1->pb, rtp_ts);
|
|
put_be32(&s1->pb, s->packet_count);
|
|
put_be32(&s1->pb, s->octet_count);
|
|
put_flush_packet(&s1->pb);
|
|
}
|
|
|
|
/* send an rtp packet. sequence number is incremented, but the caller
|
|
must update the timestamp itself */
|
|
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
|
|
#ifdef DEBUG
|
|
printf("rtp_send_data size=%d\n", len);
|
|
#endif
|
|
|
|
/* build the RTP header */
|
|
put_byte(&s1->pb, (RTP_VERSION << 6));
|
|
put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
|
|
put_be16(&s1->pb, s->seq);
|
|
put_be32(&s1->pb, s->timestamp);
|
|
put_be32(&s1->pb, s->ssrc);
|
|
|
|
put_buffer(&s1->pb, buf1, len);
|
|
put_flush_packet(&s1->pb);
|
|
|
|
s->seq++;
|
|
s->octet_count += len;
|
|
s->packet_count++;
|
|
}
|
|
|
|
/* send an integer number of samples and compute time stamp and fill
|
|
the rtp send buffer before sending. */
|
|
static void rtp_send_samples(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size, int sample_size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
int len, max_packet_size, n;
|
|
|
|
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
|
|
/* not needed, but who nows */
|
|
if ((size % sample_size) != 0)
|
|
av_abort();
|
|
n = 0;
|
|
while (size > 0) {
|
|
s->buf_ptr = s->buf;
|
|
len = FFMIN(max_packet_size, size);
|
|
|
|
/* copy data */
|
|
memcpy(s->buf_ptr, buf1, len);
|
|
s->buf_ptr += len;
|
|
buf1 += len;
|
|
size -= len;
|
|
s->timestamp = s->cur_timestamp + n / sample_size;
|
|
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
|
|
n += (s->buf_ptr - s->buf);
|
|
}
|
|
}
|
|
|
|
/* NOTE: we suppose that exactly one frame is given as argument here */
|
|
/* XXX: test it */
|
|
static void rtp_send_mpegaudio(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
int len, count, max_packet_size;
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
/* test if we must flush because not enough space */
|
|
len = (s->buf_ptr - s->buf);
|
|
if ((len + size) > max_packet_size) {
|
|
if (len > 4) {
|
|
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
|
|
s->buf_ptr = s->buf + 4;
|
|
}
|
|
}
|
|
if (s->buf_ptr == s->buf + 4) {
|
|
s->timestamp = s->cur_timestamp;
|
|
}
|
|
|
|
/* add the packet */
|
|
if (size > max_packet_size) {
|
|
/* big packet: fragment */
|
|
count = 0;
|
|
while (size > 0) {
|
|
len = max_packet_size - 4;
|
|
if (len > size)
|
|
len = size;
|
|
/* build fragmented packet */
|
|
s->buf[0] = 0;
|
|
s->buf[1] = 0;
|
|
s->buf[2] = count >> 8;
|
|
s->buf[3] = count;
|
|
memcpy(s->buf + 4, buf1, len);
|
|
ff_rtp_send_data(s1, s->buf, len + 4, 0);
|
|
size -= len;
|
|
buf1 += len;
|
|
count += len;
|
|
}
|
|
} else {
|
|
if (s->buf_ptr == s->buf + 4) {
|
|
/* no fragmentation possible */
|
|
s->buf[0] = 0;
|
|
s->buf[1] = 0;
|
|
s->buf[2] = 0;
|
|
s->buf[3] = 0;
|
|
}
|
|
memcpy(s->buf_ptr, buf1, size);
|
|
s->buf_ptr += size;
|
|
}
|
|
}
|
|
|
|
static void rtp_send_raw(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
int len, max_packet_size;
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
while (size > 0) {
|
|
len = max_packet_size;
|
|
if (len > size)
|
|
len = size;
|
|
|
|
s->timestamp = s->cur_timestamp;
|
|
ff_rtp_send_data(s1, buf1, len, (len == size));
|
|
|
|
buf1 += len;
|
|
size -= len;
|
|
}
|
|
}
|
|
|
|
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
|
|
static void rtp_send_mpegts_raw(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
int len, out_len;
|
|
|
|
while (size >= TS_PACKET_SIZE) {
|
|
len = s->max_payload_size - (s->buf_ptr - s->buf);
|
|
if (len > size)
|
|
len = size;
|
|
memcpy(s->buf_ptr, buf1, len);
|
|
buf1 += len;
|
|
size -= len;
|
|
s->buf_ptr += len;
|
|
|
|
out_len = s->buf_ptr - s->buf;
|
|
if (out_len >= s->max_payload_size) {
|
|
ff_rtp_send_data(s1, s->buf, out_len, 0);
|
|
s->buf_ptr = s->buf;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* write an RTP packet. 'buf1' must contain a single specific frame. */
|
|
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int rtcp_bytes;
|
|
int size= pkt->size;
|
|
uint8_t *buf1= pkt->data;
|
|
|
|
#ifdef DEBUG
|
|
printf("%d: write len=%d\n", pkt->stream_index, size);
|
|
#endif
|
|
|
|
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
|
|
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
|
|
RTCP_TX_RATIO_DEN;
|
|
if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
|
|
(av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
|
|
rtcp_send_sr(s1, av_gettime());
|
|
s->last_octet_count = s->octet_count;
|
|
s->first_packet = 0;
|
|
}
|
|
s->cur_timestamp = s->base_timestamp + pkt->pts;
|
|
|
|
switch(st->codec->codec_id) {
|
|
case CODEC_ID_PCM_MULAW:
|
|
case CODEC_ID_PCM_ALAW:
|
|
case CODEC_ID_PCM_U8:
|
|
case CODEC_ID_PCM_S8:
|
|
rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
|
|
break;
|
|
case CODEC_ID_PCM_U16BE:
|
|
case CODEC_ID_PCM_U16LE:
|
|
case CODEC_ID_PCM_S16BE:
|
|
case CODEC_ID_PCM_S16LE:
|
|
rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
|
|
break;
|
|
case CODEC_ID_MP2:
|
|
case CODEC_ID_MP3:
|
|
rtp_send_mpegaudio(s1, buf1, size);
|
|
break;
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
ff_rtp_send_mpegvideo(s1, buf1, size);
|
|
break;
|
|
case CODEC_ID_AAC:
|
|
ff_rtp_send_aac(s1, buf1, size);
|
|
break;
|
|
case CODEC_ID_MPEG2TS:
|
|
rtp_send_mpegts_raw(s1, buf1, size);
|
|
break;
|
|
default:
|
|
/* better than nothing : send the codec raw data */
|
|
rtp_send_raw(s1, buf1, size);
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat rtp_muxer = {
|
|
"rtp",
|
|
"RTP output format",
|
|
NULL,
|
|
NULL,
|
|
sizeof(RTPDemuxContext),
|
|
CODEC_ID_PCM_MULAW,
|
|
CODEC_ID_NONE,
|
|
rtp_write_header,
|
|
rtp_write_packet,
|
|
};
|