2008-01-04 20:09:48 +00:00
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/*
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* RTP output format
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2009-01-19 15:46:40 +00:00
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* Copyright (c) 2002 Fabrice Bellard
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2008-01-04 20:09:48 +00:00
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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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2008-05-09 11:56:36 +00:00
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2008-01-04 20:09:48 +00:00
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#include "avformat.h"
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#include "mpegts.h"
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2010-03-10 22:21:39 +00:00
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#include "internal.h"
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2011-06-04 11:58:23 +00:00
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#include "libavutil/mathematics.h"
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2010-03-28 08:48:45 +00:00
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#include "libavutil/random_seed.h"
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2011-05-18 12:00:03 +00:00
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#include "libavutil/opt.h"
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2008-01-04 20:09:48 +00:00
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2009-02-06 10:35:52 +00:00
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#include "rtpenc.h"
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2008-01-04 20:09:48 +00:00
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//#define DEBUG
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2011-05-18 12:00:03 +00:00
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static const AVOption options[] = {
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2012-02-23 09:38:24 +00:00
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FF_RTP_FLAG_OPTS(RTPMuxContext, flags)
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2011-10-04 05:38:01 +00:00
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{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
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2012-02-23 09:56:15 +00:00
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{ "max_packet_size", "Max packet size", offsetof(RTPMuxContext, max_packet_size), AV_OPT_TYPE_INT, {.dbl = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
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2011-05-18 12:00:03 +00:00
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{ NULL },
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};
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static const AVClass rtp_muxer_class = {
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.class_name = "RTP muxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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2008-01-04 20:09:48 +00:00
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#define RTCP_SR_SIZE 28
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2009-03-27 21:36:44 +00:00
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static int is_supported(enum CodecID id)
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{
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switch(id) {
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2009-04-07 06:41:55 +00:00
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case CODEC_ID_H263:
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case CODEC_ID_H263P:
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2009-03-27 21:36:44 +00:00
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case CODEC_ID_H264:
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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case CODEC_ID_MPEG4:
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case CODEC_ID_AAC:
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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case CODEC_ID_PCM_ALAW:
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case CODEC_ID_PCM_MULAW:
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case CODEC_ID_PCM_S8:
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case CODEC_ID_PCM_S16BE:
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case CODEC_ID_PCM_S16LE:
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case CODEC_ID_PCM_U16BE:
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case CODEC_ID_PCM_U16LE:
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case CODEC_ID_PCM_U8:
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case CODEC_ID_MPEG2TS:
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2009-04-08 07:16:14 +00:00
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case CODEC_ID_AMR_NB:
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case CODEC_ID_AMR_WB:
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2010-08-07 11:16:07 +00:00
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case CODEC_ID_VORBIS:
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case CODEC_ID_THEORA:
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2010-08-16 14:21:17 +00:00
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case CODEC_ID_VP8:
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2010-09-15 17:35:39 +00:00
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case CODEC_ID_ADPCM_G722:
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2011-11-30 21:11:50 +00:00
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case CODEC_ID_ADPCM_G726:
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2009-03-27 21:36:44 +00:00
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return 1;
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default:
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return 0;
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}
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}
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2008-01-04 20:09:48 +00:00
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static int rtp_write_header(AVFormatContext *s1)
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{
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2009-02-06 10:35:52 +00:00
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RTPMuxContext *s = s1->priv_data;
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2012-02-23 09:54:13 +00:00
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int n;
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2008-01-04 20:09:48 +00:00
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AVStream *st;
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2010-09-04 22:10:54 +00:00
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if (s1->nb_streams != 1) {
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av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
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return AVERROR(EINVAL);
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}
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2008-01-04 20:09:48 +00:00
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st = s1->streams[0];
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2009-03-27 21:36:44 +00:00
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if (!is_supported(st->codec->codec_id)) {
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2011-08-17 14:17:47 +00:00
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av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
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2009-03-27 21:36:44 +00:00
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return -1;
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}
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2008-01-04 20:09:48 +00:00
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2011-09-26 15:56:48 +00:00
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if (s->payload_type < 0)
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s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
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2010-05-23 08:53:40 +00:00
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s->base_timestamp = av_get_random_seed();
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2008-01-04 20:09:48 +00:00
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s->timestamp = s->base_timestamp;
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s->cur_timestamp = 0;
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2010-05-23 08:53:40 +00:00
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s->ssrc = av_get_random_seed();
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2008-01-04 20:09:48 +00:00
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s->first_packet = 1;
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2010-03-10 22:21:39 +00:00
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s->first_rtcp_ntp_time = ff_ntp_time();
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2010-03-15 10:37:12 +00:00
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if (s1->start_time_realtime)
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/* Round the NTP time to whole milliseconds. */
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s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
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NTP_OFFSET_US;
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2008-01-04 20:09:48 +00:00
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2012-02-23 09:56:15 +00:00
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if (s->max_packet_size) {
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if (s1->pb->max_packet_size)
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s->max_packet_size = FFMIN(s->max_payload_size,
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s1->pb->max_packet_size);
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} else
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s->max_packet_size = s1->pb->max_packet_size;
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2012-02-23 09:54:13 +00:00
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if (s->max_packet_size <= 12) {
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av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s->max_packet_size);
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2008-01-04 20:09:48 +00:00
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return AVERROR(EIO);
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2012-02-09 21:28:01 +00:00
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}
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2012-02-23 09:54:13 +00:00
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s->buf = av_malloc(s->max_packet_size);
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2009-02-06 20:41:15 +00:00
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if (s->buf == NULL) {
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return AVERROR(ENOMEM);
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}
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2012-02-23 09:54:13 +00:00
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s->max_payload_size = s->max_packet_size - 12;
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2008-01-04 20:09:48 +00:00
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s->max_frames_per_packet = 0;
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if (s1->max_delay) {
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2010-03-30 23:30:55 +00:00
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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2008-01-04 20:09:48 +00:00
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if (st->codec->frame_size == 0) {
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av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
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} else {
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2011-11-30 20:03:43 +00:00
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s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN);
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2008-01-04 20:09:48 +00:00
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}
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}
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2010-03-30 23:30:55 +00:00
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if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
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2008-01-04 20:09:48 +00:00
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/* FIXME: We should round down here... */
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2008-06-25 11:55:57 +00:00
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s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
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2008-01-04 20:09:48 +00:00
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}
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}
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2011-11-29 18:28:15 +00:00
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avpriv_set_pts_info(st, 32, 1, 90000);
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2008-01-04 20:09:48 +00:00
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switch(st->codec->codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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s->buf_ptr = s->buf + 4;
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break;
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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break;
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case CODEC_ID_MPEG2TS:
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n = s->max_payload_size / TS_PACKET_SIZE;
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if (n < 1)
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n = 1;
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s->max_payload_size = n * TS_PACKET_SIZE;
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s->buf_ptr = s->buf;
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break;
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2010-06-16 12:59:47 +00:00
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case CODEC_ID_H264:
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/* check for H.264 MP4 syntax */
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2010-06-17 10:30:52 +00:00
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if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
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2010-06-16 12:59:47 +00:00
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s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
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}
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break;
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2010-08-07 11:16:07 +00:00
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case CODEC_ID_VORBIS:
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case CODEC_ID_THEORA:
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if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
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s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
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s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
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s->num_frames = 0;
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goto defaultcase;
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2010-08-16 14:21:17 +00:00
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case CODEC_ID_VP8:
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2010-12-05 11:45:57 +00:00
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av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
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"incompatible with the latest spec drafts.\n");
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2010-08-16 14:21:17 +00:00
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break;
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2010-09-15 17:35:39 +00:00
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case CODEC_ID_ADPCM_G722:
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/* Due to a historical error, the clock rate for G722 in RTP is
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* 8000, even if the sample rate is 16000. See RFC 3551. */
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2011-11-29 18:28:15 +00:00
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avpriv_set_pts_info(st, 32, 1, 8000);
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2010-09-15 17:35:39 +00:00
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break;
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2009-04-08 07:16:14 +00:00
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case CODEC_ID_AMR_NB:
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case CODEC_ID_AMR_WB:
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if (!s->max_frames_per_packet)
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s->max_frames_per_packet = 12;
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if (st->codec->codec_id == CODEC_ID_AMR_NB)
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n = 31;
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else
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n = 61;
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/* max_header_toc_size + the largest AMR payload must fit */
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if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
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av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
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return -1;
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}
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if (st->codec->channels != 1) {
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av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
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return -1;
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}
|
2008-01-04 20:09:48 +00:00
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case CODEC_ID_AAC:
|
2009-02-05 14:53:25 +00:00
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s->num_frames = 0;
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2008-01-04 20:09:48 +00:00
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default:
|
2010-08-07 11:16:07 +00:00
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defaultcase:
|
2010-03-30 23:30:55 +00:00
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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2011-11-29 18:28:15 +00:00
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avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
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2008-01-04 20:09:48 +00:00
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}
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s->buf_ptr = s->buf;
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break;
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}
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return 0;
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}
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/* send an rtcp sender report packet */
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static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
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{
|
2009-02-06 10:35:52 +00:00
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RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
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uint32_t rtp_ts;
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2011-01-29 16:46:18 +00:00
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av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
|
2008-01-04 20:09:48 +00:00
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s->last_rtcp_ntp_time = ntp_time;
|
2008-06-25 11:55:57 +00:00
|
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rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
|
2008-01-04 20:09:48 +00:00
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s1->streams[0]->time_base) + s->base_timestamp;
|
2011-02-21 18:28:17 +00:00
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avio_w8(s1->pb, (RTP_VERSION << 6));
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avio_w8(s1->pb, RTCP_SR);
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avio_wb16(s1->pb, 6); /* length in words - 1 */
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avio_wb32(s1->pb, s->ssrc);
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avio_wb32(s1->pb, ntp_time / 1000000);
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avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
|
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avio_wb32(s1->pb, rtp_ts);
|
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avio_wb32(s1->pb, s->packet_count);
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avio_wb32(s1->pb, s->octet_count);
|
2011-03-14 19:39:06 +00:00
|
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avio_flush(s1->pb);
|
2008-01-04 20:09:48 +00:00
|
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}
|
|
|
|
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|
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/* send an rtp packet. sequence number is incremented, but the caller
|
|
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|
must update the timestamp itself */
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|
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void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
|
|
|
|
{
|
2009-02-06 10:35:52 +00:00
|
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|
RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
|
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|
|
2011-01-29 16:46:18 +00:00
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av_dlog(s1, "rtp_send_data size=%d\n", len);
|
2008-01-04 20:09:48 +00:00
|
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|
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|
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/* build the RTP header */
|
2011-02-21 18:28:17 +00:00
|
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|
avio_w8(s1->pb, (RTP_VERSION << 6));
|
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|
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avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
|
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|
|
avio_wb16(s1->pb, s->seq);
|
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|
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avio_wb32(s1->pb, s->timestamp);
|
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avio_wb32(s1->pb, s->ssrc);
|
2008-01-04 20:09:48 +00:00
|
|
|
|
2011-02-21 18:28:17 +00:00
|
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avio_write(s1->pb, buf1, len);
|
2011-03-14 19:39:06 +00:00
|
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avio_flush(s1->pb);
|
2008-01-04 20:09:48 +00:00
|
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|
|
|
|
|
s->seq++;
|
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|
|
s->octet_count += len;
|
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|
|
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,
|
2011-11-30 21:10:54 +00:00
|
|
|
const uint8_t *buf1, int size, int sample_size_bits)
|
2008-01-04 20:09:48 +00:00
|
|
|
{
|
2009-02-06 10:35:52 +00:00
|
|
|
RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
|
|
|
int len, max_packet_size, n;
|
2011-11-30 21:10:54 +00:00
|
|
|
/* Calculate the number of bytes to get samples aligned on a byte border */
|
|
|
|
int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
|
2008-01-04 20:09:48 +00:00
|
|
|
|
2011-11-30 21:10:54 +00:00
|
|
|
max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
|
|
|
|
/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
|
|
|
|
if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
|
2008-01-04 20:09:48 +00:00
|
|
|
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;
|
2011-11-30 21:10:54 +00:00
|
|
|
s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
|
2008-01-04 20:09:48 +00:00
|
|
|
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
|
|
|
|
n += (s->buf_ptr - s->buf);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static void rtp_send_mpegaudio(AVFormatContext *s1,
|
|
|
|
const uint8_t *buf1, int size)
|
|
|
|
{
|
2009-02-06 10:35:52 +00:00
|
|
|
RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
|
|
|
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)
|
|
|
|
{
|
2009-02-06 10:35:52 +00:00
|
|
|
RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
|
|
|
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)
|
|
|
|
{
|
2009-02-06 10:35:52 +00:00
|
|
|
RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
|
|
|
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;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
|
|
|
{
|
2009-02-06 10:35:52 +00:00
|
|
|
RTPMuxContext *s = s1->priv_data;
|
2008-01-04 20:09:48 +00:00
|
|
|
AVStream *st = s1->streams[0];
|
|
|
|
int rtcp_bytes;
|
|
|
|
int size= pkt->size;
|
|
|
|
|
2011-01-29 16:46:18 +00:00
|
|
|
av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
|
2008-01-04 20:09:48 +00:00
|
|
|
|
|
|
|
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
|
|
|
|
RTCP_TX_RATIO_DEN;
|
2012-02-09 21:28:15 +00:00
|
|
|
if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
|
|
|
|
(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
|
|
|
|
!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
|
2010-03-10 22:21:39 +00:00
|
|
|
rtcp_send_sr(s1, ff_ntp_time());
|
2008-01-04 20:09:48 +00:00
|
|
|
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:
|
2011-11-30 21:10:54 +00:00
|
|
|
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
|
|
|
case CODEC_ID_PCM_U16BE:
|
|
|
|
case CODEC_ID_PCM_U16LE:
|
|
|
|
case CODEC_ID_PCM_S16BE:
|
|
|
|
case CODEC_ID_PCM_S16LE:
|
2011-11-30 21:10:54 +00:00
|
|
|
rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
2010-09-15 17:35:39 +00:00
|
|
|
case CODEC_ID_ADPCM_G722:
|
|
|
|
/* The actual sample size is half a byte per sample, but since the
|
|
|
|
* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
|
2011-11-30 21:10:54 +00:00
|
|
|
* the correct parameter for send_samples_bits is 8 bits per stream
|
|
|
|
* clock. */
|
|
|
|
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
|
2010-09-15 17:35:39 +00:00
|
|
|
break;
|
2011-11-30 21:11:50 +00:00
|
|
|
case CODEC_ID_ADPCM_G726:
|
|
|
|
rtp_send_samples(s1, pkt->data, size,
|
|
|
|
st->codec->bits_per_coded_sample * st->codec->channels);
|
2010-09-15 17:35:39 +00:00
|
|
|
break;
|
2008-01-04 20:09:48 +00:00
|
|
|
case CODEC_ID_MP2:
|
|
|
|
case CODEC_ID_MP3:
|
2009-12-28 13:41:09 +00:00
|
|
|
rtp_send_mpegaudio(s1, pkt->data, size);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
|
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
|
|
case CODEC_ID_MPEG2VIDEO:
|
2009-12-28 13:41:09 +00:00
|
|
|
ff_rtp_send_mpegvideo(s1, pkt->data, size);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
|
|
|
case CODEC_ID_AAC:
|
2011-05-18 12:00:03 +00:00
|
|
|
if (s->flags & FF_RTP_FLAG_MP4A_LATM)
|
2011-05-18 13:21:48 +00:00
|
|
|
ff_rtp_send_latm(s1, pkt->data, size);
|
|
|
|
else
|
|
|
|
ff_rtp_send_aac(s1, pkt->data, size);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
2009-04-08 07:16:14 +00:00
|
|
|
case CODEC_ID_AMR_NB:
|
|
|
|
case CODEC_ID_AMR_WB:
|
2009-12-28 13:41:09 +00:00
|
|
|
ff_rtp_send_amr(s1, pkt->data, size);
|
2009-04-08 07:16:14 +00:00
|
|
|
break;
|
2008-01-04 20:09:48 +00:00
|
|
|
case CODEC_ID_MPEG2TS:
|
2009-12-28 13:41:09 +00:00
|
|
|
rtp_send_mpegts_raw(s1, pkt->data, size);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
2008-01-21 14:01:57 +00:00
|
|
|
case CODEC_ID_H264:
|
2009-12-28 13:41:09 +00:00
|
|
|
ff_rtp_send_h264(s1, pkt->data, size);
|
2008-01-21 14:01:57 +00:00
|
|
|
break;
|
2009-04-07 06:41:55 +00:00
|
|
|
case CODEC_ID_H263:
|
2012-02-07 14:39:14 +00:00
|
|
|
if (s->flags & FF_RTP_FLAG_RFC2190) {
|
|
|
|
ff_rtp_send_h263_rfc2190(s1, pkt->data, size);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
/* Fallthrough */
|
2009-04-07 06:41:55 +00:00
|
|
|
case CODEC_ID_H263P:
|
2009-12-28 13:41:09 +00:00
|
|
|
ff_rtp_send_h263(s1, pkt->data, size);
|
2009-04-07 06:41:55 +00:00
|
|
|
break;
|
2010-08-07 11:16:07 +00:00
|
|
|
case CODEC_ID_VORBIS:
|
|
|
|
case CODEC_ID_THEORA:
|
|
|
|
ff_rtp_send_xiph(s1, pkt->data, size);
|
|
|
|
break;
|
2010-08-16 14:21:17 +00:00
|
|
|
case CODEC_ID_VP8:
|
|
|
|
ff_rtp_send_vp8(s1, pkt->data, size);
|
|
|
|
break;
|
2008-01-04 20:09:48 +00:00
|
|
|
default:
|
|
|
|
/* better than nothing : send the codec raw data */
|
2009-12-28 13:41:09 +00:00
|
|
|
rtp_send_raw(s1, pkt->data, size);
|
2008-01-04 20:09:48 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2009-02-06 20:41:15 +00:00
|
|
|
static int rtp_write_trailer(AVFormatContext *s1)
|
|
|
|
{
|
|
|
|
RTPMuxContext *s = s1->priv_data;
|
|
|
|
|
|
|
|
av_freep(&s->buf);
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2011-01-25 22:03:28 +00:00
|
|
|
AVOutputFormat ff_rtp_muxer = {
|
2011-07-16 20:18:12 +00:00
|
|
|
.name = "rtp",
|
|
|
|
.long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
|
|
|
|
.priv_data_size = sizeof(RTPMuxContext),
|
|
|
|
.audio_codec = CODEC_ID_PCM_MULAW,
|
2011-10-17 18:11:07 +00:00
|
|
|
.video_codec = CODEC_ID_MPEG4,
|
2011-07-16 20:18:12 +00:00
|
|
|
.write_header = rtp_write_header,
|
|
|
|
.write_packet = rtp_write_packet,
|
|
|
|
.write_trailer = rtp_write_trailer,
|
2011-05-18 12:00:03 +00:00
|
|
|
.priv_class = &rtp_muxer_class,
|
2008-01-04 20:09:48 +00:00
|
|
|
};
|