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Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit into transcode2

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# Conflicts:
#	CMakeLists.txt
#	conf/config.ini
#	src/Common/MediaSink.cpp
#	src/Common/MediaSink.h
#	src/Common/MediaSource.cpp
#	src/Common/MultiMediaSourceMuxer.h
#	src/Common/config.cpp
#	src/Common/config.h
#	src/Extension/AAC.cpp
#	src/Extension/AAC.h
#	src/Rtsp/RtpCodec.h
#	src/Rtsp/RtspMuxer.cpp
#	src/Rtsp/RtspMuxer.h
#	webrtc/Nack.cpp
#	webrtc/WebRtcTransport.cpp
This commit is contained in:
cqm
2024-07-12 09:48:01 +08:00
parent 9bf48567b2 3e7e1a317e
commit 2ffdfab71e
465 changed files with 18124 additions and 9008 deletions
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309
webrtc/WebRtcTransport.cpp
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@@ -1,16 +1,19 @@
/*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* 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 <iostream>
#include <srtp2/srtp.h>
#include "Util/base64.h"
#include "Network/sockutil.h"
#include "Common/config.h"
#include "Nack.h"
#include "RtpExt.h"
#include "Rtcp/Rtcp.h"
#include "Rtcp/RtcpFCI.h"
@@ -22,7 +25,6 @@
#include "WebRtcEchoTest.h"
#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
#include "Rtsp/RtspMediaSourceImp.h"
#define RTP_SSRC_OFFSET 1
@@ -30,7 +32,6 @@
#define RTP_CNAME "zlmediakit-rtp"
#define RTP_LABEL "zlmediakit-label"
#define RTP_MSLABEL "zlmediakit-mslabel"
#define RTP_MSID RTP_MSLABEL " " RTP_LABEL
using namespace std;
@@ -50,15 +51,28 @@ const string kTranscodeG711 = RTC_FIELD "transcodeG711";
// webrtc单端口udp服务器
const string kPort = RTC_FIELD "port";
const string kTcpPort = RTC_FIELD "tcpPort";
// 比特率设置
const string kStartBitrate = RTC_FIELD "start_bitrate";
const string kMaxBitrate = RTC_FIELD "max_bitrate";
const string kMinBitrate = RTC_FIELD "min_bitrate";
// 数据通道设置
const string kDataChannelEcho = RTC_FIELD "datachannel_echo";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
mINI::Instance()[kRembBitRate] = 0;
mINI::Instance()[kPort] = 0;
mINI::Instance()[kTcpPort] = 0;
mINI::Instance()[kPort] = 8000;
mINI::Instance()[kTcpPort] = 8000;
mINI::Instance()[kStartBitrate] = 0;
mINI::Instance()[kMaxBitrate] = 0;
mINI::Instance()[kMinBitrate] = 0;
mINI::Instance()[kDataChannelEcho] = true;
mINI::Instance()[kTranscodeG711] = 0;
});
@@ -81,6 +95,28 @@ static void translateIPFromEnv(std::vector<std::string> &v) {
}
}
static std::string getServerPrefix() {
//stun_user_name格式: base64(ip+udp_port+tcp_port) + _ + number
//其中ip为二进制char[4], udp_port/tcp_port为大端 uint16.
//number为自增长数，确保短时间内唯一
GET_CONFIG(uint16_t, udp_port, Rtc::kPort);
GET_CONFIG(uint16_t, tcp_port, Rtc::kTcpPort);
char buf[8];
auto host = SockUtil::get_local_ip();
auto addr = SockUtil::make_sockaddr(host.data(), udp_port);
//拷贝ipv4地址
memcpy(buf, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_addr), 4);
//拷贝udp端口
memcpy(buf + 4, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_port), 2);
//tcp端口转大端模式
addr = SockUtil::make_sockaddr(host.data(), tcp_port);
//拷贝tcp端口
memcpy(buf + 6, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_port), 2);
auto ret = encodeBase64(string(buf, 8)) + '_';
InfoL << "MediaServer(" << host << ":" << udp_port << ":" << tcp_port << ") prefix: " << ret;
return ret;
}
const char* sockTypeStr(Session* session) {
if (session) {
switch (session->getSock()->sockType()) {
@@ -94,7 +130,8 @@ const char* sockTypeStr(Session* session) {
WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
_poller = poller;
_identifier = "zlm_" + to_string(++s_key);
static auto prefix = getServerPrefix();
_identifier = prefix + to_string(++s_key);
_packet_pool.setSize(64);
}
@@ -173,9 +210,26 @@ void WebRtcTransport::OnDtlsTransportConnected(
onStartWebRTC();
}
#pragma pack(push, 1)
struct DtlsHeader {
uint8_t content_type;
uint16_t dtls_version;
uint16_t epoch;
uint8_t seq[6];
uint16_t length;
uint8_t payload[1];
};
#pragma pack(pop)
void WebRtcTransport::OnDtlsTransportSendData(
const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
sendSockData((char *)data, len, nullptr);
size_t offset = 0;
while(offset < len) {
auto *header = reinterpret_cast<const DtlsHeader *>(data + offset);
auto length = ntohs(header->length) + offsetof(DtlsHeader, payload);
sendSockData((char *)data + offset, length, nullptr);
offset += length;
}
}
void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
@@ -205,22 +259,47 @@ void WebRtcTransport::OnDtlsTransportApplicationDataReceived(
#ifdef ENABLE_SCTP
void WebRtcTransport::OnSctpAssociationConnecting(RTC::SctpAssociation *sctpAssociation) {
TraceL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpConnecting, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationConnected(RTC::SctpAssociation *sctpAssociation) {
InfoL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpConnected, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationFailed(RTC::SctpAssociation *sctpAssociation) {
WarnL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpFailed, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationClosed(RTC::SctpAssociation *sctpAssociation) {
InfoL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpClosed, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationSendData(
RTC::SctpAssociation *sctpAssociation, const uint8_t *data, size_t len) {
try {
NOTICE_EMIT(BroadcastRtcSctpSendArgs, Broadcast::kBroadcastRtcSctpSend, *this, data, len);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
_dtls_transport->SendApplicationData(data, len);
}
@@ -229,10 +308,33 @@ void WebRtcTransport::OnSctpAssociationMessageReceived(
InfoL << getIdentifier() << " " << streamId << " " << ppid << " " << len << " " << string((char *)msg, len);
RTC::SctpStreamParameters params;
params.streamId = streamId;
// 回显数据
_sctp->SendSctpMessage(params, ppid, msg, len);
GET_CONFIG(bool, datachannel_echo, Rtc::kDataChannelEcho);
if (datachannel_echo) {
// 回显数据
_sctp->SendSctpMessage(params, ppid, msg, len);
}
try {
NOTICE_EMIT(BroadcastRtcSctpReceivedArgs, Broadcast::kBroadcastRtcSctpReceived, *this, streamId, ppid, msg, len);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
#endif
void WebRtcTransport::sendDatachannel(uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
#ifdef ENABLE_SCTP
if (_sctp) {
RTC::SctpStreamParameters params;
params.streamId = streamId;
_sctp->SendSctpMessage(params, ppid, (uint8_t *)msg, len);
}
#else
WarnL << "WebRTC datachannel disabled!";
#endif
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple) {
@@ -242,7 +344,7 @@ void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTu
}
Session::Ptr WebRtcTransport::getSession() const {
auto tuple = _ice_server->GetSelectedTuple(true);
auto tuple = _ice_server ? _ice_server->GetSelectedTuple(true) : nullptr;
return tuple ? static_pointer_cast<Session>(tuple->shared_from_this()) : nullptr;
}
@@ -281,11 +383,31 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
}
void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
// 开启remb后关闭twcc，因为开启twcc后remb无效
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
configure.enableTWCC(!remb_bit_rate);
}
static void setSdpBitrate(RtcSession &sdp) {
GET_CONFIG(size_t, max_bitrate, Rtc::kMaxBitrate);
GET_CONFIG(size_t, min_bitrate, Rtc::kMinBitrate);
GET_CONFIG(size_t, start_bitrate, Rtc::kStartBitrate);
auto m = (RtcMedia *)(sdp.getMedia(TrackType::TrackVideo));
if (m) {
auto &plan = m->plan[0];
if (max_bitrate) plan.fmtp["x-google-max-bitrate"] = std::to_string(max_bitrate);
if (min_bitrate) plan.fmtp["x-google-min-bitrate"] = std::to_string(min_bitrate);
if (start_bitrate) plan.fmtp["x-google-start-bitrate"] = std::to_string(start_bitrate);
}
}
std::string WebRtcTransport::getAnswerSdp(const string &offer) {
try {
//// 解析offer sdp ////
@@ -296,17 +418,13 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer) {
setRemoteDtlsFingerprint(*_offer_sdp);
//// sdp 配置 ////
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
RtcConfigure configure;
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
onRtcConfigure(configure);
//// 生成answer sdp ////
_answer_sdp = configure.createAnswer(*_offer_sdp);
onCheckSdp(SdpType::answer, *_answer_sdp);
setSdpBitrate(*_answer_sdp);
_answer_sdp->checkValid();
return _answer_sdp->toString();
} catch (exception &ex) {
@@ -319,10 +437,6 @@ static bool isDtls(char *buf) {
return ((*buf > 19) && (*buf < 64));
}
static string getPeerAddress(RTC::TransportTuple *tuple) {
return tuple->get_peer_ip();
}
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
@@ -339,7 +453,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
}
if (isRtp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
WarnL << "received rtp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
@@ -349,7 +463,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
}
if (isRtcp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
WarnL << "received rtcp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
@@ -421,8 +535,7 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
#endif
}
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp)
: WebRtcTransport(poller), _preferred_tcp(preferred_tcp) {
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
InfoL << getIdentifier();
}
@@ -562,14 +675,16 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
});
for (auto &m : sdp.media) {
m.addr.reset();
m.addr.address = extern_ips.empty() ? SockUtil::get_local_ip() : extern_ips[0];
m.addr.address = extern_ips.empty() ? _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip : extern_ips[0];
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
GET_CONFIG(uint16_t, udp_port, Rtc::kPort);
GET_CONFIG(uint16_t, tcp_port, Rtc::kTcpPort);
m.rtcp_addr.port = udp_port ? udp_port : tcp_port;
m.port = m.rtcp_addr.port;
m.port = m.port ? (udp_port ? udp_port : tcp_port) : 0;
if (m.type != TrackApplication) {
m.rtcp_addr.port = m.port;
}
sdp.origin.address = m.addr.address;
}
@@ -592,9 +707,9 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
// 发送的ssrc我们随便定义，因为在发送rtp时会修改为此值
ssrc.ssrc = m.type + RTP_SSRC_OFFSET;
ssrc.cname = RTP_CNAME;
ssrc.label = RTP_LABEL;
ssrc.label = std::string(RTP_LABEL) + '-' + m.mid;
ssrc.mslabel = RTP_MSLABEL;
ssrc.msid = RTP_MSID;
ssrc.msid = ssrc.mslabel + ' ' + ssrc.label;
if (m.getRelatedRtxPlan(m.plan[0].pt)) {
// rtx ssrc
@@ -625,7 +740,7 @@ makeIceCandidate(std::string ip, uint16_t port, uint32_t priority = 100, std::st
candidate->foundation = proto + "candidate";
// 优先级，单candidate时随便
candidate->priority = priority;
candidate->address = ip;
candidate->address = std::move(ip);
candidate->port = port;
candidate->type = "host";
if (proto == "tcp") {
@@ -636,6 +751,12 @@ makeIceCandidate(std::string ip, uint16_t port, uint32_t priority = 100, std::st
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
if (!_cands.empty()) {
for (auto &cand : _cands) {
configure.addCandidate(cand);
}
return;
}
GET_CONFIG(uint16_t, local_udp_port, Rtc::kPort);
GET_CONFIG(uint16_t, local_tcp_port, Rtc::kTcpPort);
@@ -649,7 +770,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
return ret;
});
if (extern_ips.empty()) {
std::string local_ip = SockUtil::get_local_ip();
std::string local_ip = _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip;
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
} else {
@@ -663,6 +784,18 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
}
}
void WebRtcTransportImp::setPreferredTcp(bool flag) {
_preferred_tcp = flag;
}
void WebRtcTransportImp::setLocalIp(std::string local_ip) {
_local_ip = std::move(local_ip);
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
}
///////////////////////////////////////////////////////////////////
class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {
@@ -672,12 +805,12 @@ public:
_on_nack = std::move(on_nack);
setOnSorted(std::move(cb));
//设置jitter buffer参数
RtpTrackImp::setParams(1024, NackContext::kNackMaxMS, 512);
GET_CONFIG(uint32_t, nack_maxms, Rtc::kNackMaxMS);
GET_CONFIG(uint32_t, nack_max_rtp, Rtc::kNackMaxSize);
RtpTrackImp::setParams(nack_max_rtp, nack_maxms, nack_max_rtp / 2);
_nack_ctx.setOnNack([this](const FCI_NACK &nack) { onNack(nack); });
}
~RtpChannel() override = default;
RtpPacket::Ptr inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx) {
auto rtp = RtpTrack::inputRtp(type, sample_rate, ptr, len);
if (!rtp) {
@@ -775,7 +908,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
if (!rtp_chn) {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
} else {
// InfoL << "接收丢包率,ssrc:" << sr->ssrc << ",loss rate(%):" << rtp_chn->getLossRate();
// 设置rtp时间戳与ntp时间戳的对应关系
rtp_chn->setNtpStamp(sr->rtpts, sr->getNtpUnixStampMS());
auto rr = rtp_chn->createRtcpRR(sr, track->answer_ssrc_rtp);
@@ -1153,29 +1285,42 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
_map_creator[type] = std::move(cb);
}
std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer) {
return const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
void WebRtcPluginManager::setListener(Listener cb) {
lock_guard<mutex> lck(_mtx_creator);
_listener = std::move(cb);
}
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb) {
void WebRtcPluginManager::negotiateSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateWebRtc &cb_in) {
onCreateWebRtc cb;
lock_guard<mutex> lck(_mtx_creator);
if (_listener) {
auto listener = _listener;
auto args_ptr = args.shared_from_this();
auto sender_ptr = static_pointer_cast<Session>(sender.shared_from_this());
cb = [listener, sender_ptr, type, args_ptr, cb_in](const WebRtcInterface &rtc) {
listener(*sender_ptr, type, *args_ptr, rtc);
cb_in(rtc);
};
} else {
cb = cb_in;
}
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
cb(WebRtcException(SockException(Err_other, "the type can not supported")));
cb_in(WebRtcException(SockException(Err_other, "the type can not supported")));
return;
}
it->second(sender, args, cb);
}
void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
Broadcast::PublishAuthInvoker invoker = [cb, info, preferred_tcp](const string &err, const ProtocolOption &option) mutable {
Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@@ -1214,25 +1359,23 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
}
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, preferred_tcp);
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
push_src->setListener(rtc);
cb(*rtc);
};
// rtsp推流需要鉴权
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, static_cast<SockInfo &>(sender));
auto flag = NOTICE_EMIT(BroadcastMediaPublishArgs, Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, sender);
if (!flag) {
// 该事件无人监听,默认不鉴权
invoker("", ProtocolOption());
}
}
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
auto session_ptr = static_pointer_cast<Session>(sender.shared_from_this());
Broadcast::AuthInvoker invoker = [cb, info, session_ptr, preferred_tcp](const string &err) mutable {
Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@@ -1246,24 +1389,78 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
return;
}
// 还原成rtc，目的是为了hook时识别哪种播放协议
info.schema = RTC_SCHEMA;
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, preferred_tcp);
info.schema = "rtc";
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
cb(*rtc);
});
};
// 广播通用播放url鉴权事件
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, info, invoker, static_cast<SockInfo &>(sender));
auto flag = NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, info, invoker, sender);
if (!flag) {
// 该事件无人监听,默认不鉴权
invoker("");
}
}
static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
{
static auto is_vaild_ip = [](const std::string &ip) -> bool {
int a, b, c, d;
return sscanf(ip.c_str(), "%d.%d.%d.%d", &a, &b, &c, &d) == 4;
};
std::string host = args["Host"];
if (!host.empty()) {
auto local_ip = host.substr(0, host.find(':'));
if (!is_vaild_ip(local_ip) || local_ip == "127.0.0.1") {
local_ip = "";
}
rtc.setLocalIp(std::move(local_ip));
}
}
bool preferred_tcp = args["preferred_tcp"];
{
rtc.setPreferredTcp(preferred_tcp);
}
{
vector<SdpAttrCandidate> cands;
{
auto cand_str = trim(args["cand_udp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// udp优先
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
cands.emplace_back(std::move(*ice_cand));
}
}
{
auto cand_str = trim(args["cand_tcp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// tcp模式
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 120 : 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
rtc.setIceCandidate(std::move(cands));
}
}
}
static onceToken s_rtc_auto_register([]() {
#if !defined (NDEBUG)
// debug模式才开启echo插件
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
#endif
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
WebRtcPluginManager::Instance().setListener([](Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
});
});
}// namespace mediakit
}// namespace mediakit