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release 8.0

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夏楚
2023-12-09 16:23:51 +08:00
committed by GitHub
parent 44c82752b3
commit bbe8f4a018
346 changed files with 2529 additions and 3435 deletions
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124
webrtc/WebRtcTransport.cpp
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@@ -1,15 +1,17 @@
/*
* 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 "RtpExt.h"
#include "Rtcp/Rtcp.h"
@@ -22,7 +24,6 @@
#include "WebRtcEchoTest.h"
#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
#include "Rtsp/RtspMediaSourceImp.h"
#define RTP_SSRC_OFFSET 1
@@ -47,15 +48,23 @@ const string kExternIP = RTC_FIELD "externIP";
const string kRembBitRate = RTC_FIELD "rembBitRate";
// 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";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
mINI::Instance()[kRembBitRate] = 0;
mINI::Instance()[kPort] = 0;
mINI::Instance()[kTcpPort] = 0;
mINI::Instance()[kStartBitrate] = 0;
mINI::Instance()[kMaxBitrate] = 0;
mINI::Instance()[kMinBitrate] = 0;
});
} // namespace RTC
@@ -77,6 +86,28 @@ 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()) {
@@ -90,7 +121,8 @@ 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);
}
@@ -295,6 +327,20 @@ void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
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 ////
@@ -316,6 +362,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer) {
//// 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) {
@@ -636,7 +683,7 @@ 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") {
@@ -647,6 +694,12 @@ 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);
@@ -674,6 +727,10 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
}
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
}
///////////////////////////////////////////////////////////////////
class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {
@@ -687,8 +744,6 @@ public:
_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) {
@@ -786,7 +841,6 @@ 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);
@@ -1168,8 +1222,26 @@ std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& off
return const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
}
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb) {
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_in) {
onCreateRtc 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")));
@@ -1272,10 +1344,40 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
}
static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc) {
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()), 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()), 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
const_cast<WebRtcInterface &>(rtc).setIceCandidate(std::move(cands));
}
}
static onceToken s_rtc_auto_register([]() {
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
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) {
set_webrtc_cands(args, rtc);
});
});
}// namespace mediakit