mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-25 12:11:36 +08:00
2ead272187
rtsp直接代理时会从config frame生成rtp,在获取VideoTrack配置帧列表时存在线程安全风险; 同时简化getConfigFrames函数代码,去除缓存逻辑。
164 lines
5.9 KiB
C++
164 lines
5.9 KiB
C++
/*
|
|
* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
|
|
*
|
|
* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
|
|
*
|
|
* Use of this source code is governed by MIT-like license that can be found in the
|
|
* LICENSE file in the root of the source tree. All contributing project authors
|
|
* may be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "WebRtcPlayer.h"
|
|
|
|
#include "Common/config.h"
|
|
#include "Extension/Factory.h"
|
|
#include "Util/base64.h"
|
|
|
|
using namespace std;
|
|
|
|
namespace mediakit {
|
|
|
|
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
|
|
const RtspMediaSource::Ptr &src,
|
|
const MediaInfo &info) {
|
|
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
|
|
ptr->onDestory();
|
|
delete ptr;
|
|
});
|
|
ret->onCreate();
|
|
return ret;
|
|
}
|
|
|
|
WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
|
|
const RtspMediaSource::Ptr &src,
|
|
const MediaInfo &info) : WebRtcTransportImp(poller) {
|
|
_media_info = info;
|
|
_play_src = src;
|
|
CHECK(src);
|
|
|
|
GET_CONFIG(bool, direct_proxy, Rtsp::kDirectProxy);
|
|
_send_config_frames_once = direct_proxy;
|
|
}
|
|
|
|
void WebRtcPlayer::onStartWebRTC() {
|
|
auto playSrc = _play_src.lock();
|
|
if(!playSrc){
|
|
onShutdown(SockException(Err_shutdown, "rtsp media source was shutdown"));
|
|
return ;
|
|
}
|
|
WebRtcTransportImp::onStartWebRTC();
|
|
if (canSendRtp()) {
|
|
playSrc->pause(false);
|
|
_reader = playSrc->getRing()->attach(getPoller(), true);
|
|
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
|
|
weak_ptr<Session> weak_session = static_pointer_cast<Session>(getSession());
|
|
_reader->setGetInfoCB([weak_session]() {
|
|
Any ret;
|
|
ret.set(static_pointer_cast<SockInfo>(weak_session.lock()));
|
|
return ret;
|
|
});
|
|
_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
|
|
auto strong_self = weak_self.lock();
|
|
if (!strong_self) {
|
|
return;
|
|
}
|
|
|
|
if (strong_self->_send_config_frames_once && !pkt->empty()) {
|
|
const auto &first_rtp = pkt->front();
|
|
strong_self->sendConfigFrames(first_rtp->getSeq(), first_rtp->sample_rate, first_rtp->getStamp(), first_rtp->ntp_stamp);
|
|
strong_self->_send_config_frames_once = false;
|
|
}
|
|
|
|
size_t i = 0;
|
|
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
|
|
//TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
|
|
strong_self->onSendRtp(rtp, ++i == pkt->size());
|
|
});
|
|
});
|
|
_reader->setDetachCB([weak_self]() {
|
|
auto strong_self = weak_self.lock();
|
|
if (!strong_self) {
|
|
return;
|
|
}
|
|
strong_self->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
|
|
});
|
|
|
|
_reader->setMessageCB([weak_self] (const toolkit::Any &data) {
|
|
auto strong_self = weak_self.lock();
|
|
if (!strong_self) {
|
|
return;
|
|
}
|
|
if (data.is<Buffer>()) {
|
|
auto &buffer = data.get<Buffer>();
|
|
// PPID 51: 文本string
|
|
// PPID 53: 二进制
|
|
strong_self->sendDatachannel(0, 51, buffer.data(), buffer.size());
|
|
} else {
|
|
WarnL << "Send unknown message type to webrtc player: " << data.type_name();
|
|
}
|
|
});
|
|
}
|
|
}
|
|
void WebRtcPlayer::onDestory() {
|
|
auto duration = getDuration();
|
|
auto bytes_usage = getBytesUsage();
|
|
//流量统计事件广播
|
|
GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
|
|
if (_reader && getSession()) {
|
|
WarnL << "RTC播放器(" << _media_info.shortUrl() << ")结束播放,耗时(s):" << duration;
|
|
if (bytes_usage >= iFlowThreshold * 1024) {
|
|
NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration, true, *getSession());
|
|
}
|
|
}
|
|
WebRtcTransportImp::onDestory();
|
|
}
|
|
|
|
void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
|
|
auto playSrc = _play_src.lock();
|
|
if(!playSrc){
|
|
return ;
|
|
}
|
|
WebRtcTransportImp::onRtcConfigure(configure);
|
|
//这是播放
|
|
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
|
|
configure.setPlayRtspInfo(playSrc->getSdp());
|
|
}
|
|
|
|
void WebRtcPlayer::sendConfigFrames(uint32_t before_seq, uint32_t sample_rate, uint32_t timestamp, uint64_t ntp_timestamp) {
|
|
auto play_src = _play_src.lock();
|
|
if (!play_src) {
|
|
return;
|
|
}
|
|
SdpParser parser(play_src->getSdp());
|
|
auto video_sdp = parser.getTrack(TrackVideo);
|
|
if (!video_sdp) {
|
|
return;
|
|
}
|
|
auto video_track = dynamic_pointer_cast<VideoTrack>(Factory::getTrackBySdp(video_sdp));
|
|
if (!video_track) {
|
|
return;
|
|
}
|
|
auto frames = video_track->getConfigFrames();
|
|
if (frames.empty()) {
|
|
return;
|
|
}
|
|
auto encoder = mediakit::Factory::getRtpEncoderByCodecId(video_track->getCodecId(), 0);
|
|
if (!encoder) {
|
|
return;
|
|
}
|
|
|
|
GET_CONFIG(uint32_t, video_mtu, Rtp::kVideoMtuSize);
|
|
encoder->setRtpInfo(0, video_mtu, sample_rate, 0, 0, 0);
|
|
|
|
auto seq = before_seq - frames.size();
|
|
for (const auto &frame : frames) {
|
|
auto rtp = encoder->getRtpInfo().makeRtp(TrackVideo, frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize(), false, 0);
|
|
auto header = rtp->getHeader();
|
|
header->seq = htons(seq++);
|
|
header->stamp = htonl(timestamp);
|
|
rtp->ntp_stamp = ntp_timestamp;
|
|
onSendRtp(rtp, false);
|
|
}
|
|
}
|
|
|
|
}// namespace mediakit
|