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647 lines
22 KiB
647 lines
22 KiB
/* |
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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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|
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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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|
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#include "rtpenc.h" |
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|
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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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|
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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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|
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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_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_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_ADPCM_G722: |
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case AV_CODEC_ID_ADPCM_G726: |
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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; |
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AVStream *st; |
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|
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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->codec->codec_id)) { |
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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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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->codec, -1); |
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|
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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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|
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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 %d 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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s->max_frames_per_packet = 0; |
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if (s1->max_delay > 0) { |
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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int frame_size = av_get_audio_frame_duration(st->codec, 0); |
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if (!frame_size) |
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frame_size = st->codec->frame_size; |
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if (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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s->max_frames_per_packet = |
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av_rescale_q_rnd(s1->max_delay, |
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AV_TIME_BASE_Q, |
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(AVRational){ frame_size, st->codec->sample_rate }, |
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AV_ROUND_DOWN); |
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} |
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} |
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if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { |
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/* FIXME: We should round down here... */ |
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if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) { |
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s->max_frames_per_packet = av_rescale_q(s1->max_delay, |
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(AVRational){1, 1000000}, |
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av_inv_q(st->avg_frame_rate)); |
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} else |
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s->max_frames_per_packet = 1; |
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} |
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} |
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avpriv_set_pts_info(st, 32, 1, 90000); |
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switch(st->codec->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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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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s->buf_ptr = s->buf; |
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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->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { |
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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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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/H264 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->codec->extradata_size > 21 && st->codec->extradata[0] == 1) { |
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s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1; |
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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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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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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->codec->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->codec->block_align != 38 && st->codec->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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if (!s->max_frames_per_packet) |
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s->max_frames_per_packet = 1; |
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s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, |
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s->max_payload_size / st->codec->block_align); |
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goto defaultcase; |
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case AV_CODEC_ID_AMR_NB: |
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case AV_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 == 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->codec->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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case AV_CODEC_ID_AAC: |
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s->num_frames = 0; |
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default: |
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defaultcase: |
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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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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fail: |
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av_freep(&s->buf); |
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return AVERROR(EINVAL); |
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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_dlog(s1, "RTCP: %02x %"PRIx64" %x\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_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time)); |
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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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|
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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_dlog(s1, "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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|
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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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|
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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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|
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max_packet_size = s->max_payload_size; |
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|
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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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|
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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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} |
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|
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static void rtp_send_raw(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, max_packet_size; |
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|
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max_packet_size = s->max_payload_size; |
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|
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while (size > 0) { |
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len = max_packet_size; |
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if (len > size) |
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len = size; |
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|
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s->timestamp = s->cur_timestamp; |
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ff_rtp_send_data(s1, buf1, len, (len == size)); |
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|
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buf1 += len; |
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size -= len; |
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} |
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} |
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|
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/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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static void rtp_send_mpegts_raw(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, out_len; |
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|
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s->timestamp = s->cur_timestamp; |
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while (size >= TS_PACKET_SIZE) { |
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len = s->max_payload_size - (s->buf_ptr - s->buf); |
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if (len > size) |
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len = size; |
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memcpy(s->buf_ptr, buf1, len); |
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buf1 += len; |
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size -= len; |
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s->buf_ptr += len; |
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|
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out_len = s->buf_ptr - s->buf; |
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if (out_len >= s->max_payload_size) { |
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ff_rtp_send_data(s1, s->buf, out_len, 0); |
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s->buf_ptr = s->buf; |
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} |
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} |
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} |
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|
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static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size) |
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{ |
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RTPMuxContext *s = s1->priv_data; |
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AVStream *st = s1->streams[0]; |
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int frame_duration = av_get_audio_frame_duration(st->codec, 0); |
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int frame_size = st->codec->block_align; |
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int frames = size / frame_size; |
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|
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while (frames > 0) { |
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int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames); |
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|
|
if (!s->num_frames) { |
|
s->buf_ptr = s->buf; |
|
s->timestamp = s->cur_timestamp; |
|
} |
|
memcpy(s->buf_ptr, buf, n * frame_size); |
|
frames -= n; |
|
s->num_frames += n; |
|
s->buf_ptr += n * frame_size; |
|
buf += n * frame_size; |
|
s->cur_timestamp += n * 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_dlog(s1, "%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->codec->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->codec->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->codec->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->codec->channels); |
|
case AV_CODEC_ID_ADPCM_G726: |
|
return rtp_send_samples(s1, pkt->data, size, |
|
st->codec->bits_per_coded_sample * st->codec->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_H264: |
|
ff_rtp_send_h264(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 = 0; |
|
const uint8_t *mb_info = |
|
av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO, |
|
&mb_info_size); |
|
if (!mb_info) { |
|
av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n"); |
|
return AVERROR(ENOMEM); |
|
} |
|
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_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_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, |
|
};
|
|
|