984b914c55
This makes the packetization spec compliant for cases where one single GOB doesn't fit into an RTP packet. Signed-off-by: Martin Storsjö <martin@martin.st>
502 lines
16 KiB
C
502 lines
16 KiB
C
/*
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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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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//#define DEBUG
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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, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
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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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{ 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 CodecID id)
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{
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switch(id) {
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case CODEC_ID_H263:
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case CODEC_ID_H263P:
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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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case CODEC_ID_AMR_NB:
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case CODEC_ID_AMR_WB:
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case CODEC_ID_VORBIS:
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case CODEC_ID_THEORA:
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case CODEC_ID_VP8:
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case CODEC_ID_ADPCM_G722:
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case CODEC_ID_ADPCM_G726:
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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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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 %x\n", 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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s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
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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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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)
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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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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_packet_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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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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return AVERROR(EIO);
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}
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s->buf = av_malloc(s->max_packet_size);
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if (s->buf == NULL) {
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return AVERROR(ENOMEM);
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}
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s->max_payload_size = s->max_packet_size - 12;
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s->max_frames_per_packet = 0;
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if (s1->max_delay) {
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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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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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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}
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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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s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
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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 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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case 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 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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case CODEC_ID_VP8:
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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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break;
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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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avpriv_set_pts_info(st, 32, 1, 8000);
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break;
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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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}
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case 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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}
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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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{
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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_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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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_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++;
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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 void 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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av_abort();
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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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}
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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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}
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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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max_packet_size = s->max_payload_size;
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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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s->timestamp = s->cur_timestamp;
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ff_rtp_send_data(s1, buf1, len, (len == size));
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buf1 += len;
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size -= len;
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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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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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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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static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
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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 rtcp_bytes;
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int size= pkt->size;
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av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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RTCP_TX_RATIO_DEN;
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if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
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(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
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!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
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rtcp_send_sr(s1, ff_ntp_time());
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s->last_octet_count = s->octet_count;
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s->first_packet = 0;
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}
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s->cur_timestamp = s->base_timestamp + pkt->pts;
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switch(st->codec->codec_id) {
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case CODEC_ID_PCM_MULAW:
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case CODEC_ID_PCM_ALAW:
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case CODEC_ID_PCM_U8:
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case CODEC_ID_PCM_S8:
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rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
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break;
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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_S16BE:
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case CODEC_ID_PCM_S16LE:
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rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
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break;
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case CODEC_ID_ADPCM_G722:
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/* The actual sample size is half a byte per sample, but since the
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* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
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* the correct parameter for send_samples_bits is 8 bits per stream
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* clock. */
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rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
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break;
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case CODEC_ID_ADPCM_G726:
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rtp_send_samples(s1, pkt->data, size,
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st->codec->bits_per_coded_sample * st->codec->channels);
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break;
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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rtp_send_mpegaudio(s1, pkt->data, size);
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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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ff_rtp_send_mpegvideo(s1, pkt->data, size);
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break;
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case CODEC_ID_AAC:
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if (s->flags & FF_RTP_FLAG_MP4A_LATM)
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ff_rtp_send_latm(s1, pkt->data, size);
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else
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ff_rtp_send_aac(s1, pkt->data, size);
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break;
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case CODEC_ID_AMR_NB:
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case CODEC_ID_AMR_WB:
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ff_rtp_send_amr(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_MPEG2TS:
|
|
rtp_send_mpegts_raw(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_H264:
|
|
ff_rtp_send_h264(s1, pkt->data, size);
|
|
break;
|
|
case 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);
|
|
ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
|
|
break;
|
|
}
|
|
/* Fallthrough */
|
|
case CODEC_ID_H263P:
|
|
ff_rtp_send_h263(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_VORBIS:
|
|
case CODEC_ID_THEORA:
|
|
ff_rtp_send_xiph(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_VP8:
|
|
ff_rtp_send_vp8(s1, pkt->data, size);
|
|
break;
|
|
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;
|
|
|
|
av_freep(&s->buf);
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat ff_rtp_muxer = {
|
|
.name = "rtp",
|
|
.long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
|
|
.priv_data_size = sizeof(RTPMuxContext),
|
|
.audio_codec = CODEC_ID_PCM_MULAW,
|
|
.video_codec = CODEC_ID_MPEG4,
|
|
.write_header = rtp_write_header,
|
|
.write_packet = rtp_write_packet,
|
|
.write_trailer = rtp_write_trailer,
|
|
.priv_class = &rtp_muxer_class,
|
|
};
|