mirror of
https://github.com/pineappleEA/pineapple-src.git
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663 lines
22 KiB
C
Executable File
663 lines
22 KiB
C
Executable File
/*
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* RTP 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 "internal.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/random_seed.h"
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#include "libavutil/opt.h"
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#include "rtpenc.h"
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static const AVOption options[] = {
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FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
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{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
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{ "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
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{ "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
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{ "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
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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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#define RTCP_SR_SIZE 28
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static int is_supported(enum AVCodecID id)
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{
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switch(id) {
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case AV_CODEC_ID_DIRAC:
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case AV_CODEC_ID_H261:
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case AV_CODEC_ID_H263:
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case AV_CODEC_ID_H263P:
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case AV_CODEC_ID_H264:
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case AV_CODEC_ID_HEVC:
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case AV_CODEC_ID_MPEG1VIDEO:
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case AV_CODEC_ID_MPEG2VIDEO:
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case AV_CODEC_ID_MPEG4:
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case AV_CODEC_ID_AAC:
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case AV_CODEC_ID_MP2:
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case AV_CODEC_ID_MP3:
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case AV_CODEC_ID_PCM_ALAW:
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case AV_CODEC_ID_PCM_MULAW:
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case AV_CODEC_ID_PCM_S8:
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case AV_CODEC_ID_PCM_S16BE:
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case AV_CODEC_ID_PCM_S16LE:
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case AV_CODEC_ID_PCM_S24BE:
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case AV_CODEC_ID_PCM_U16BE:
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case AV_CODEC_ID_PCM_U16LE:
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case AV_CODEC_ID_PCM_U8:
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case AV_CODEC_ID_MPEG2TS:
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case AV_CODEC_ID_AMR_NB:
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case AV_CODEC_ID_AMR_WB:
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case AV_CODEC_ID_VORBIS:
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case AV_CODEC_ID_THEORA:
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case AV_CODEC_ID_VP8:
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case AV_CODEC_ID_VP9:
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case AV_CODEC_ID_ADPCM_G722:
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case AV_CODEC_ID_ADPCM_G726:
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case AV_CODEC_ID_ADPCM_G726LE:
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case AV_CODEC_ID_ILBC:
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case AV_CODEC_ID_MJPEG:
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case AV_CODEC_ID_SPEEX:
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case AV_CODEC_ID_OPUS:
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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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static int rtp_write_header(AVFormatContext *s1)
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{
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RTPMuxContext *s = s1->priv_data;
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int n, ret = AVERROR(EINVAL);
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AVStream *st;
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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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st = s1->streams[0];
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if (!is_supported(st->codecpar->codec_id)) {
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av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
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return -1;
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}
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if (s->payload_type < 0) {
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/* Re-validate non-dynamic payload types */
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if (st->id < RTP_PT_PRIVATE)
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st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
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s->payload_type = st->id;
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} else {
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/* private option takes priority */
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st->id = s->payload_type;
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}
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s->base_timestamp = av_get_random_seed();
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s->timestamp = s->base_timestamp;
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s->cur_timestamp = 0;
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if (!s->ssrc)
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s->ssrc = av_get_random_seed();
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s->first_packet = 1;
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s->first_rtcp_ntp_time = ff_ntp_time();
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if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
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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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// Pick a random sequence start number, but in the lower end of the
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// available range, so that any wraparound doesn't happen immediately.
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// (Immediate wraparound would be an issue for SRTP.)
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if (s->seq < 0) {
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if (s1->flags & AVFMT_FLAG_BITEXACT) {
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s->seq = 0;
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} else
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s->seq = av_get_random_seed() & 0x0fff;
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} else
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s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
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if (s1->packet_size) {
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if (s1->pb->max_packet_size)
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s1->packet_size = FFMIN(s1->packet_size,
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s1->pb->max_packet_size);
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} else
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s1->packet_size = s1->pb->max_packet_size;
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if (s1->packet_size <= 12) {
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av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
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return AVERROR(EIO);
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}
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s->buf = av_malloc(s1->packet_size);
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if (!s->buf) {
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return AVERROR(ENOMEM);
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}
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s->max_payload_size = s1->packet_size - 12;
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if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
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} else {
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avpriv_set_pts_info(st, 32, 1, 90000);
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}
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s->buf_ptr = s->buf;
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switch(st->codecpar->codec_id) {
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case AV_CODEC_ID_MP2:
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case AV_CODEC_ID_MP3:
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s->buf_ptr = s->buf + 4;
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avpriv_set_pts_info(st, 32, 1, 90000);
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break;
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case AV_CODEC_ID_MPEG1VIDEO:
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case AV_CODEC_ID_MPEG2VIDEO:
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break;
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case AV_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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break;
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case AV_CODEC_ID_DIRAC:
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if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
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av_log(s, AV_LOG_ERROR,
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"Packetizing VC-2 is experimental and does not use all values "
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"of the specification "
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"(even though most receivers may handle it just fine). "
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"Please set -strict experimental in order to enable it.\n");
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ret = AVERROR_EXPERIMENTAL;
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goto fail;
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}
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break;
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case AV_CODEC_ID_H261:
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if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
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av_log(s, AV_LOG_ERROR,
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"Packetizing H.261 is experimental and produces incorrect "
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"packetization for cases where GOBs don't fit into packets "
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"(even though most receivers may handle it just fine). "
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"Please set -f_strict experimental in order to enable it.\n");
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ret = AVERROR_EXPERIMENTAL;
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goto fail;
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}
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break;
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case AV_CODEC_ID_H264:
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/* check for H.264 MP4 syntax */
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if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
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s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
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}
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break;
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case AV_CODEC_ID_HEVC:
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/* Only check for the standardized hvcC version of extradata, keeping
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* things simple and similar to the avcC/H.264 case above, instead
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* of trying to handle the pre-standardization versions (as in
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* libavcodec/hevc.c). */
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if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
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s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
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}
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break;
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case AV_CODEC_ID_VP9:
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if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
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av_log(s, AV_LOG_ERROR,
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"Packetizing VP9 is experimental and its specification is "
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"still in draft state. "
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"Please set -strict experimental in order to enable it.\n");
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ret = AVERROR_EXPERIMENTAL;
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goto fail;
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}
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break;
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case AV_CODEC_ID_VORBIS:
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case AV_CODEC_ID_THEORA:
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s->max_frames_per_packet = 15;
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break;
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case AV_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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avpriv_set_pts_info(st, 32, 1, 8000);
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break;
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case AV_CODEC_ID_OPUS:
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if (st->codecpar->channels > 2) {
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av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
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goto fail;
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}
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/* The opus RTP RFC says that all opus streams should use 48000 Hz
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* as clock rate, since all opus sample rates can be expressed in
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* this clock rate, and sample rate changes on the fly are supported. */
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avpriv_set_pts_info(st, 32, 1, 48000);
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break;
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case AV_CODEC_ID_ILBC:
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if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
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av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
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goto fail;
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}
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s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
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break;
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case AV_CODEC_ID_AMR_NB:
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case AV_CODEC_ID_AMR_WB:
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s->max_frames_per_packet = 50;
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if (st->codecpar->codec_id == AV_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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goto fail;
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}
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if (st->codecpar->channels != 1) {
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av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
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goto fail;
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}
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break;
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case AV_CODEC_ID_AAC:
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s->max_frames_per_packet = 50;
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break;
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default:
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break;
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}
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return 0;
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fail:
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av_freep(&s->buf);
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return ret;
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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, int bye)
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{
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RTPMuxContext *s = s1->priv_data;
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uint32_t rtp_ts;
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av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
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s->last_rtcp_ntp_time = ntp_time;
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rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
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s1->streams[0]->time_base) + s->base_timestamp;
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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);
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if (s->cname) {
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int len = FFMIN(strlen(s->cname), 255);
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avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
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avio_w8(s1->pb, RTCP_SDES);
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avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
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avio_wb32(s1->pb, s->ssrc);
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avio_w8(s1->pb, 0x01); /* CNAME */
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avio_w8(s1->pb, len);
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avio_write(s1->pb, s->cname, len);
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avio_w8(s1->pb, 0); /* END */
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for (len = (7 + len) % 4; len % 4; len++)
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avio_w8(s1->pb, 0);
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}
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if (bye) {
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avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
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avio_w8(s1->pb, RTCP_BYE);
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avio_wb16(s1->pb, 1); /* length in words - 1 */
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avio_wb32(s1->pb, s->ssrc);
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}
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avio_flush(s1->pb);
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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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void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
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{
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RTPMuxContext *s = s1->priv_data;
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av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
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/* build the RTP header */
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avio_w8(s1->pb, RTP_VERSION << 6);
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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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avio_wb32(s1->pb, s->timestamp);
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avio_wb32(s1->pb, s->ssrc);
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avio_write(s1->pb, buf1, len);
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avio_flush(s1->pb);
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s->seq = (s->seq + 1) & 0xffff;
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s->octet_count += len;
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s->packet_count++;
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}
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/* send an integer number of samples and compute time stamp and fill
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the rtp send buffer before sending. */
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static int rtp_send_samples(AVFormatContext *s1,
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const uint8_t *buf1, int size, int sample_size_bits)
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{
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RTPMuxContext *s = s1->priv_data;
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int len, max_packet_size, n;
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/* Calculate the number of bytes to get samples aligned on a byte border */
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int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
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max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
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/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
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if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
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return AVERROR(EINVAL);
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n = 0;
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while (size > 0) {
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s->buf_ptr = s->buf;
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len = FFMIN(max_packet_size, size);
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/* copy data */
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memcpy(s->buf_ptr, buf1, len);
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s->buf_ptr += len;
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buf1 += len;
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size -= len;
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s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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n += (s->buf_ptr - s->buf);
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}
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return 0;
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}
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static void rtp_send_mpegaudio(AVFormatContext *s1,
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const uint8_t *buf1, int size)
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{
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RTPMuxContext *s = s1->priv_data;
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int len, count, max_packet_size;
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max_packet_size = s->max_payload_size;
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/* test if we must flush because not enough space */
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len = (s->buf_ptr - s->buf);
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if ((len + size) > max_packet_size) {
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if (len > 4) {
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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s->buf_ptr = s->buf + 4;
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}
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}
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if (s->buf_ptr == s->buf + 4) {
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s->timestamp = s->cur_timestamp;
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}
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/* add the packet */
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if (size > max_packet_size) {
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/* big packet: fragment */
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count = 0;
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while (size > 0) {
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len = max_packet_size - 4;
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if (len > size)
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len = size;
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/* build fragmented packet */
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s->buf[0] = 0;
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s->buf[1] = 0;
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s->buf[2] = count >> 8;
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s->buf[3] = count;
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memcpy(s->buf + 4, buf1, len);
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ff_rtp_send_data(s1, s->buf, len + 4, 0);
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size -= len;
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buf1 += len;
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count += len;
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}
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} else {
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if (s->buf_ptr == s->buf + 4) {
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/* no fragmentation possible */
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s->buf[0] = 0;
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s->buf[1] = 0;
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s->buf[2] = 0;
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s->buf[3] = 0;
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}
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memcpy(s->buf_ptr, buf1, size);
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s->buf_ptr += size;
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}
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|
}
|
|
|
|
static void rtp_send_raw(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPMuxContext *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)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
int len, out_len;
|
|
|
|
s->timestamp = s->cur_timestamp;
|
|
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_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
|
|
int frame_size = st->codecpar->block_align;
|
|
int frames = size / frame_size;
|
|
|
|
while (frames > 0) {
|
|
if (s->num_frames > 0 &&
|
|
av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
|
|
s1->max_delay, AV_TIME_BASE_Q) >= 0) {
|
|
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
|
|
s->num_frames = 0;
|
|
}
|
|
|
|
if (!s->num_frames) {
|
|
s->buf_ptr = s->buf;
|
|
s->timestamp = s->cur_timestamp;
|
|
}
|
|
memcpy(s->buf_ptr, buf, frame_size);
|
|
frames--;
|
|
s->num_frames++;
|
|
s->buf_ptr += frame_size;
|
|
buf += frame_size;
|
|
s->cur_timestamp += frame_duration;
|
|
|
|
if (s->num_frames == s->max_frames_per_packet) {
|
|
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
|
|
s->num_frames = 0;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int rtcp_bytes;
|
|
int size= pkt->size;
|
|
|
|
av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
|
|
|
|
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) &&
|
|
(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
|
|
!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
|
|
rtcp_send_sr(s1, ff_ntp_time(), 0);
|
|
s->last_octet_count = s->octet_count;
|
|
s->first_packet = 0;
|
|
}
|
|
s->cur_timestamp = s->base_timestamp + pkt->pts;
|
|
|
|
switch(st->codecpar->codec_id) {
|
|
case AV_CODEC_ID_PCM_MULAW:
|
|
case AV_CODEC_ID_PCM_ALAW:
|
|
case AV_CODEC_ID_PCM_U8:
|
|
case AV_CODEC_ID_PCM_S8:
|
|
return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
|
|
case AV_CODEC_ID_PCM_U16BE:
|
|
case AV_CODEC_ID_PCM_U16LE:
|
|
case AV_CODEC_ID_PCM_S16BE:
|
|
case AV_CODEC_ID_PCM_S16LE:
|
|
return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
|
|
case AV_CODEC_ID_PCM_S24BE:
|
|
return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->channels);
|
|
case AV_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,
|
|
* the correct parameter for send_samples_bits is 8 bits per stream
|
|
* clock. */
|
|
return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
|
|
case AV_CODEC_ID_ADPCM_G726:
|
|
case AV_CODEC_ID_ADPCM_G726LE:
|
|
return rtp_send_samples(s1, pkt->data, size,
|
|
st->codecpar->bits_per_coded_sample * st->codecpar->channels);
|
|
case AV_CODEC_ID_MP2:
|
|
case AV_CODEC_ID_MP3:
|
|
rtp_send_mpegaudio(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_MPEG1VIDEO:
|
|
case AV_CODEC_ID_MPEG2VIDEO:
|
|
ff_rtp_send_mpegvideo(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_AAC:
|
|
if (s->flags & FF_RTP_FLAG_MP4A_LATM)
|
|
ff_rtp_send_latm(s1, pkt->data, size);
|
|
else
|
|
ff_rtp_send_aac(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_AMR_NB:
|
|
case AV_CODEC_ID_AMR_WB:
|
|
ff_rtp_send_amr(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_MPEG2TS:
|
|
rtp_send_mpegts_raw(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_DIRAC:
|
|
ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
|
|
break;
|
|
case AV_CODEC_ID_H264:
|
|
ff_rtp_send_h264_hevc(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_H261:
|
|
ff_rtp_send_h261(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_H263:
|
|
if (s->flags & FF_RTP_FLAG_RFC2190) {
|
|
int mb_info_size;
|
|
const uint8_t *mb_info =
|
|
av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
|
|
&mb_info_size);
|
|
ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
|
|
break;
|
|
}
|
|
/* Fallthrough */
|
|
case AV_CODEC_ID_H263P:
|
|
ff_rtp_send_h263(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_HEVC:
|
|
ff_rtp_send_h264_hevc(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_VORBIS:
|
|
case AV_CODEC_ID_THEORA:
|
|
ff_rtp_send_xiph(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_VP8:
|
|
ff_rtp_send_vp8(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_VP9:
|
|
ff_rtp_send_vp9(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_ILBC:
|
|
rtp_send_ilbc(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_MJPEG:
|
|
ff_rtp_send_jpeg(s1, pkt->data, size);
|
|
break;
|
|
case AV_CODEC_ID_OPUS:
|
|
if (size > s->max_payload_size) {
|
|
av_log(s1, AV_LOG_ERROR,
|
|
"Packet size %d too large for max RTP payload size %d\n",
|
|
size, s->max_payload_size);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
/* Intentional fallthrough */
|
|
default:
|
|
/* better than nothing : send the codec raw data */
|
|
rtp_send_raw(s1, pkt->data, size);
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_write_trailer(AVFormatContext *s1)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
|
|
/* If the caller closes and recreates ->pb, this might actually
|
|
* be NULL here even if it was successfully allocated at the start. */
|
|
if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
|
|
rtcp_send_sr(s1, ff_ntp_time(), 1);
|
|
av_freep(&s->buf);
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat ff_rtp_muxer = {
|
|
.name = "rtp",
|
|
.long_name = NULL_IF_CONFIG_SMALL("RTP output"),
|
|
.priv_data_size = sizeof(RTPMuxContext),
|
|
.audio_codec = AV_CODEC_ID_PCM_MULAW,
|
|
.video_codec = AV_CODEC_ID_MPEG4,
|
|
.write_header = rtp_write_header,
|
|
.write_packet = rtp_write_packet,
|
|
.write_trailer = rtp_write_trailer,
|
|
.priv_class = &rtp_muxer_class,
|
|
.flags = AVFMT_TS_NONSTRICT,
|
|
};
|