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ZLMediaKit
Commit 0b355759 by ziyue
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整理webrtc相关代码命名空间

parent 15affeff
正在显示 包含 170 行增加 和 112 行删除
api/source/mk_webrtc_private.h
......@@ -24,6 +24,8 @@
#include "../webrtc/WebRtcPusher.h"
#include "../webrtc/WebRtcTransport.h"
using namespace mediakit;
namespace API {
typedef enum {
NotFound = -500, //未找到
......
server/main.cpp
......@@ -286,7 +286,7 @@ int start_main(int argc,char *argv[]) {
}
return Socket::createSocket(new_poller, false);
});
uint16_t rtcPort = mINI::Instance()[RTC::kPort];
uint16_t rtcPort = mINI::Instance()[Rtc::kPort];
#endif//defined(ENABLE_WEBRTC)
......
tests/test_server.cpp
......@@ -83,7 +83,7 @@ onceToken token1([](){
} // namespace mediakit
#define REALM "realm_zlmedaikit"
#define REALM "realm_zlmediakit"
static map<string,FlvRecorder::Ptr> s_mapFlvRecorder;
static mutex s_mtxFlvRecorder;
......
webrtc/Nack.cpp
......@@ -12,7 +12,8 @@
using namespace std;
using namespace toolkit;
using namespace mediakit;
namespace mediakit {
static constexpr uint32_t kMaxNackMS = 5 * 1000;
static constexpr uint32_t kRtpCacheCheckInterval = 100;
......@@ -92,7 +93,6 @@ int64_t NackList::getRtpStamp(uint16_t seq) {
return it->second->getStampMS(false);
}
////////////////////////////////////////////////////////////////////////////////////////////////
void NackContext::received(uint16_t seq, bool is_rtx) {
......@@ -101,7 +101,7 @@ void NackContext::received(uint16_t seq, bool is_rtx) {
}
if (is_rtx || (seq < _last_max_seq && !(seq < 1024 && _last_max_seq > UINT16_MAX - 1024))) {
//重传包或
//seq回退，且非回环，那么这个应该是重传包
// seq回退，且非回环，那么这个应该是重传包
onRtx(seq);
return;
}
......@@ -127,7 +127,7 @@ void NackContext::received(uint16_t seq, bool is_rtx) {
_seq.clear();
_last_max_seq = max_seq;
} else {
//seq不连续，有丢包
// seq不连续，有丢包
if (min_seq == _last_max_seq + 1) {
//前面部分seq是连续的，未丢包，移除之
eraseFrontSeq();
......@@ -135,7 +135,7 @@ void NackContext::received(uint16_t seq, bool is_rtx) {
//有丢包，丢包从_last_max_seq开始
auto nack_rtp_count = FCI_NACK::kBitSize;
if (max_seq > nack_rtp_count + _last_max_seq) {
if (max_seq > nack_rtp_count + _last_max_seq) {
vector<bool> vec;
vec.resize(FCI_NACK::kBitSize, false);
for (size_t i = 0; i < nack_rtp_count; ++i) {
......@@ -170,7 +170,7 @@ void NackContext::eraseFrontSeq() {
//前面部分seq是连续的，未丢包，移除之
for (auto it = _seq.begin(); it != _seq.end();) {
if (*it != _last_max_seq + 1) {
//seq不连续，丢包了
// seq不连续，丢包了
break;
}
_last_max_seq = *it;
......@@ -187,9 +187,9 @@ void NackContext::onRtx(uint16_t seq) {
_nack_send_status.erase(it);
if (rtt >= 0) {
//rtt不肯小于0
// rtt不肯小于0
_rtt = rtt;
//InfoL << "rtt:" << rtt;
// InfoL << "rtt:" << rtt;
}
}
......@@ -230,7 +230,7 @@ uint64_t NackContext::reSendNack() {
//更新nack发送时间戳
it->second.update_stamp = now;
if (++(it->second.nack_count) == kNackMaxCount) {
//nack次数太多，移除之
// nack次数太多，移除之
it = _nack_send_status.erase(it);
continue;
}
......@@ -269,3 +269,5 @@ uint64_t NackContext::reSendNack() {
//重传间隔不得低于5ms
return max(_rtt, 5);
}
} // namespace mediakit
webrtc/Nack.h
......@@ -14,38 +14,39 @@
#include "Rtsp/Rtsp.h"
#include "Rtcp/RtcpFCI.h"
namespace mediakit {
class NackList {
public:
NackList() = default;
~NackList() = default;
void pushBack(mediakit::RtpPacket::Ptr rtp);
void forEach(const mediakit::FCI_NACK &nack, const std::function<void(const mediakit::RtpPacket::Ptr &rtp)> &cb);
void pushBack(RtpPacket::Ptr rtp);
void forEach(const FCI_NACK &nack, const std::function<void(const RtpPacket::Ptr &rtp)> &cb);
private:
void popFront();
uint32_t getCacheMS();
int64_t getRtpStamp(uint16_t seq);
mediakit::RtpPacket::Ptr *getRtp(uint16_t seq);
RtpPacket::Ptr *getRtp(uint16_t seq);
private:
uint32_t _cache_ms_check = 0;
std::deque<uint16_t> _nack_cache_seq;
std::unordered_map<uint16_t, mediakit::RtpPacket::Ptr> _nack_cache_pkt;
std::unordered_map<uint16_t, RtpPacket::Ptr> _nack_cache_pkt;
};
class NackContext {
public:
using Ptr = std::shared_ptr<NackContext>;
using onNack = std::function<void(const mediakit::FCI_NACK &nack)>;
using onNack = std::function<void(const FCI_NACK &nack)>;
//最大保留的rtp丢包状态个数
static constexpr auto kNackMaxSize = 2048;
//rtp丢包状态最长保留时间
// rtp丢包状态最长保留时间
static constexpr auto kNackMaxMS = 3 * 1000;
//nack最多请求重传10次
// nack最多请求重传10次
static constexpr auto kNackMaxCount = 10;
//nack重传频率，rtt的倍数
// nack重传频率，rtt的倍数
static constexpr auto kNackIntervalRatio = 1.0f;
NackContext() = default;
......@@ -57,8 +58,8 @@ public:
private:
void eraseFrontSeq();
void doNack(const mediakit::FCI_NACK &nack, bool record_nack);
void recordNack(const mediakit::FCI_NACK &nack);
void doNack(const FCI_NACK &nack, bool record_nack);
void recordNack(const FCI_NACK &nack);
void onRtx(uint16_t seq);
private:
......@@ -67,12 +68,14 @@ private:
std::set<uint16_t> _seq;
uint16_t _last_max_seq = 0;
struct NackStatus{
struct NackStatus {
uint64_t first_stamp;
uint64_t update_stamp;
int nack_count = 0;
};
std::map<uint16_t/*seq*/, NackStatus > _nack_send_status;
std::map<uint16_t /*seq*/, NackStatus> _nack_send_status;
};
} // namespace mediakit
#endif //ZLMEDIAKIT_NACK_H
webrtc/RtpExt.cpp
......@@ -17,7 +17,8 @@
using namespace std;
using namespace toolkit;
using namespace mediakit;
namespace mediakit {
//https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
//https://tools.ietf.org/html/rfc5285
......@@ -644,3 +645,4 @@ void RtpExtContext::onGetRtp(uint8_t pt, uint32_t ssrc, const string &rid){
}
}
}// namespace mediakit
\ No newline at end of file
webrtc/RtpExt.h
......@@ -17,6 +17,7 @@
#include "Common/macros.h"
#include "Rtsp/Rtsp.h"
namespace mediakit {
#define RTP_EXT_MAP(XX) \
XX(ssrc_audio_level, "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \
......@@ -53,7 +54,7 @@ public:
friend class RtpExtContext;
~RtpExt() = default;
static std::map<uint8_t/*id*/, RtpExt/*data*/> getExtValue(const mediakit::RtpHeader *header);
static std::map<uint8_t/*id*/, RtpExt/*data*/> getExtValue(const RtpHeader *header);
static RtpExtType getExtType(const std::string &url);
static const std::string& getExtUrl(RtpExtType type);
static const char *getExtName(RtpExtType type);
......@@ -122,7 +123,7 @@ public:
void setOnGetRtp(OnGetRtp cb);
std::string getRid(uint32_t ssrc) const;
void setRid(uint32_t ssrc, const std::string &rid);
RtpExt changeRtpExtId(const mediakit::RtpHeader *header, bool is_recv, std::string *rid_ptr = nullptr, RtpExtType type = RtpExtType::padding);
RtpExt changeRtpExtId(const RtpHeader *header, bool is_recv, std::string *rid_ptr = nullptr, RtpExtType type = RtpExtType::padding);
private:
void onGetRtp(uint8_t pt, uint32_t ssrc, const std::string &rid);
......@@ -137,4 +138,5 @@ private:
std::unordered_map<uint32_t/*simulcast ssrc*/, std::string/*rid*/> _ssrc_to_rid;
};
} //namespace mediakit
#endif //ZLMEDIAKIT_RTPEXT_H
webrtc/Sdp.cpp
......@@ -14,9 +14,10 @@
using namespace std;
using namespace toolkit;
using namespace mediakit;
namespace RTC {
namespace mediakit {
namespace Rtc {
#define RTC_FIELD "rtc."
const string kPreferredCodecA = RTC_FIELD"preferredCodecA";
const string kPreferredCodecV = RTC_FIELD"preferredCodecV";
......@@ -1404,7 +1405,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
switch (type) {
case TrackAudio: {
//此处调整偏好的编码格式优先级
GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, RTC::kPreferredCodecA, toCodecArray);
GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, Rtc::kPreferredCodecA, toCodecArray);
CHECK(!s_preferred_codec.empty(), "rtc音频偏好codec不能为空");
preferred_codec = s_preferred_codec;
......@@ -1423,7 +1424,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
}
case TrackVideo: {
//此处调整偏好的编码格式优先级
GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, RTC::kPreferredCodecV, toCodecArray);
GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, Rtc::kPreferredCodecV, toCodecArray);
CHECK(!s_preferred_codec.empty(), "rtc视频偏好codec不能为空");
preferred_codec = s_preferred_codec;
......@@ -1811,3 +1812,5 @@ void RtcConfigure::onSelectPlan(RtcCodecPlan &plan, CodecId codec) const {
plan.fmtp[kMode] = mode.empty() ? "0" : mode;
}
}
} // namespace mediakit
\ No newline at end of file
webrtc/Sdp.h
......@@ -18,6 +18,8 @@
#include "Extension/Frame.h"
#include "Common/Parser.h"
namespace mediakit {
//https://datatracker.ietf.org/doc/rfc4566/?include_text=1
//https://blog.csdn.net/aggresss/article/details/109850434
//https://aggresss.blog.csdn.net/article/details/106436703
......@@ -189,7 +191,7 @@ class SdpMedia : public SdpItem {
public:
// 5.14. Media Descriptions ("m=")
// m=<media> <port> <proto> <fmt> ...
mediakit::TrackType type;
TrackType type;
uint16_t port;
//RTP/AVP：应用场景为视频/音频的 RTP 协议。参考 RFC 3551
//RTP/SAVP：应用场景为视频/音频的 SRTP 协议。参考 RFC 3711
......@@ -374,7 +376,7 @@ class SdpAttrFmtp : public SdpItem {
public:
//fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
uint8_t pt;
std::map<std::string/*key*/, std::string/*value*/, mediakit::StrCaseCompare> fmtp;
std::map<std::string/*key*/, std::string/*value*/, StrCaseCompare> fmtp;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "fmtp";}
......@@ -600,7 +602,7 @@ public:
uint32_t channel = 0;
//rtcp反馈
std::set<std::string> rtcp_fb;
std::map<std::string/*key*/, std::string/*value*/, mediakit::StrCaseCompare> fmtp;
std::map<std::string/*key*/, std::string/*value*/, StrCaseCompare> fmtp;
std::string getFmtp(const char *key) const;
};
......@@ -608,7 +610,7 @@ public:
//rtc 媒体描述
class RtcMedia{
public:
mediakit::TrackType type{mediakit::TrackType::TrackInvalid};
TrackType type{TrackType::TrackInvalid};
std::string mid;
uint16_t port{0};
SdpConnection addr;
......@@ -675,8 +677,8 @@ public:
void checkValid() const;
std::string toString() const;
std::string toRtspSdp() const;
const RtcMedia *getMedia(mediakit::TrackType type) const;
bool supportRtcpFb(const std::string &name, mediakit::TrackType type = mediakit::TrackType::TrackVideo) const;
const RtcMedia *getMedia(TrackType type) const;
bool supportRtcpFb(const std::string &name, TrackType type = TrackType::TrackVideo) const;
bool supportSimulcast() const;
bool isOnlyDatachannel() const;
......@@ -706,10 +708,10 @@ public:
std::set<std::string> rtcp_fb;
std::map<RtpExtType, RtpDirection> extmap;
std::vector<mediakit::CodecId> preferred_codec;
std::vector<CodecId> preferred_codec;
std::vector<SdpAttrCandidate> candidate;
void setDefaultSetting(mediakit::TrackType type);
void setDefaultSetting(TrackType type);
void enableTWCC(bool enable = true);
void enableREMB(bool enable = true);
};
......@@ -719,19 +721,19 @@ public:
RtcTrackConfigure application;
void setDefaultSetting(std::string ice_ufrag, std::string ice_pwd, RtpDirection direction, const SdpAttrFingerprint &fingerprint);
void addCandidate(const SdpAttrCandidate &candidate, mediakit::TrackType type = mediakit::TrackInvalid);
void addCandidate(const SdpAttrCandidate &candidate, TrackType type = TrackInvalid);
std::shared_ptr<RtcSession> createAnswer(const RtcSession &offer) const;
void setPlayRtspInfo(const std::string &sdp);
void enableTWCC(bool enable = true, mediakit::TrackType type = mediakit::TrackInvalid);
void enableREMB(bool enable = true, mediakit::TrackType type = mediakit::TrackInvalid);
void enableTWCC(bool enable = true, TrackType type = TrackInvalid);
void enableREMB(bool enable = true, TrackType type = TrackInvalid);
private:
void matchMedia(const std::shared_ptr<RtcSession> &ret, const RtcMedia &media) const;
bool onCheckCodecProfile(const RtcCodecPlan &plan, mediakit::CodecId codec) const;
void onSelectPlan(RtcCodecPlan &plan, mediakit::CodecId codec) const;
bool onCheckCodecProfile(const RtcCodecPlan &plan, CodecId codec) const;
void onSelectPlan(RtcCodecPlan &plan, CodecId codec) const;
private:
RtcCodecPlan::Ptr _rtsp_video_plan;
......@@ -748,5 +750,6 @@ private:
~SdpConst() = delete;
};
}// namespace mediakit
#endif //ZLMEDIAKIT_SDP_H
webrtc/TwccContext.cpp
......@@ -11,7 +11,7 @@
#include "TwccContext.h"
#include "Rtcp/RtcpFCI.h"
using namespace mediakit;
namespace mediakit {
enum class ExtSeqStatus : int {
normal = 0,
......@@ -121,3 +121,5 @@ void TwccContext::clearStatus() {
void TwccContext::setOnSendTwccCB(TwccContext::onSendTwccCB cb) {
_cb = std::move(cb);
}
}// namespace mediakit
\ No newline at end of file
webrtc/TwccContext.h
......@@ -16,6 +16,8 @@
#include <functional>
#include "Util/TimeTicker.h"
namespace mediakit {
class TwccContext {
public:
using onSendTwccCB = std::function<void(uint32_t ssrc, std::string fci)>;
......@@ -44,5 +46,5 @@ private:
onSendTwccCB _cb;
};
}// namespace mediakit
#endif //ZLMEDIAKIT_TWCCCONTEXT_H
webrtc/WebRtcEchoTest.cpp
......@@ -10,6 +10,8 @@
#include "WebRtcEchoTest.h"
namespace mediakit {
WebRtcEchoTest::Ptr WebRtcEchoTest::create(const EventPoller::Ptr &poller) {
WebRtcEchoTest::Ptr ret(new WebRtcEchoTest(poller), [](WebRtcEchoTest *ptr) {
ptr->onDestory();
......@@ -48,4 +50,6 @@ void WebRtcEchoTest::onCheckSdp(SdpType type, RtcSession &sdp) {
}
}
}
}
\ No newline at end of file
}
}// namespace mediakit
\ No newline at end of file
webrtc/WebRtcEchoTest.h
......@@ -13,6 +13,8 @@
#include "WebRtcTransport.h"
namespace mediakit {
class WebRtcEchoTest : public WebRtcTransportImp {
public:
using Ptr = std::shared_ptr<WebRtcEchoTest>;
......@@ -26,7 +28,7 @@ protected:
void onRtp(const char *buf, size_t len, uint64_t stamp_ms) override;
void onRtcp(const char *buf, size_t len) override;
void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override {};
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override {};
void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};
......@@ -34,4 +36,5 @@ private:
WebRtcEchoTest(const EventPoller::Ptr &poller);
};
}// namespace mediakit
#endif //ZLMEDIAKIT_WEBRTCECHOTEST_H
webrtc/WebRtcPlayer.cpp
......@@ -11,7 +11,8 @@
#include "WebRtcPlayer.h"
using namespace std;
using namespace mediakit;
namespace mediakit {
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
......@@ -88,3 +89,5 @@ void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
configure.setPlayRtspInfo(_play_src->getSdp());
}
}// namespace mediakit
\ No newline at end of file
webrtc/WebRtcPlayer.h
......@@ -13,30 +13,32 @@
#include "WebRtcTransport.h"
namespace mediakit {
class WebRtcPlayer : public WebRtcTransportImp {
public:
using Ptr = std::shared_ptr<WebRtcPlayer>;
~WebRtcPlayer() override = default;
static Ptr create(const EventPoller::Ptr &poller, const mediakit::RtspMediaSource::Ptr &src, const mediakit::MediaInfo &info);
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
protected:
///////WebRtcTransportImp override///////
void onStartWebRTC() override;
void onDestory() override;
void onRtcConfigure(RtcConfigure &configure) const override;
void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override {};
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
private:
WebRtcPlayer(const EventPoller::Ptr &poller, const mediakit::RtspMediaSource::Ptr &src, const mediakit::MediaInfo &info);
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
private:
//媒体相关元数据
mediakit::MediaInfo _media_info;
MediaInfo _media_info;
//播放的rtsp源
mediakit::RtspMediaSource::Ptr _play_src;
RtspMediaSource::Ptr _play_src;
//播放rtsp源的reader对象
mediakit::RtspMediaSource::RingType::RingReader::Ptr _reader;
RtspMediaSource::RingType::RingReader::Ptr _reader;
};
#endif //ZLMEDIAKIT_WEBRTCPLAYER_H
}// namespace mediakit
#endif // ZLMEDIAKIT_WEBRTCPLAYER_H
webrtc/WebRtcPusher.cpp
......@@ -11,13 +11,14 @@
#include "WebRtcPusher.h"
using namespace std;
using namespace mediakit;
namespace mediakit {
WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const mediakit::ProtocolOption &option) {
const ProtocolOption &option) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
ptr->onDestory();
delete ptr;
......@@ -30,7 +31,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const mediakit::ProtocolOption &option) : WebRtcTransportImp(poller) {
const ProtocolOption &option) : WebRtcTransportImp(poller) {
_media_info = info;
_push_src = src;
_push_src_ownership = ownership;
......@@ -142,7 +143,7 @@ void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
}
float WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
float WebRtcPusher::getLossRate(MediaSource &sender,TrackType type){
return WebRtcTransportImp::getLossRate(type);
}
......@@ -155,4 +156,6 @@ void WebRtcPusher::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport
void WebRtcPusher::onRtcpBye(){
_push_src = nullptr;
WebRtcTransportImp::onRtcpBye();
}
\ No newline at end of file
}
}// namespace mediakit
\ No newline at end of file
webrtc/WebRtcPusher.h
......@@ -13,19 +13,21 @@
#include "WebRtcTransport.h"
class WebRtcPusher : public WebRtcTransportImp, public mediakit::MediaSourceEvent {
namespace mediakit {
class WebRtcPusher : public WebRtcTransportImp, public MediaSourceEvent {
public:
using Ptr = std::shared_ptr<WebRtcPusher>;
~WebRtcPusher() override = default;
static Ptr create(const EventPoller::Ptr &poller, const mediakit::RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership, const mediakit::MediaInfo &info, const mediakit::ProtocolOption &option);
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
protected:
///////WebRtcTransportImp override///////
void onStartWebRTC() override;
void onDestory() override;
void onRtcConfigure(RtcConfigure &configure) const override;
void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override;
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override;
void onRtcpBye() override;
//// dtls相关的回调 ////
void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
......@@ -33,35 +35,36 @@ protected:
protected:
///////MediaSourceEvent override///////
// 关闭
bool close(mediakit::MediaSource &sender) override;
bool close(MediaSource &sender) override;
// 播放总人数
int totalReaderCount(mediakit::MediaSource &sender) override;
int totalReaderCount(MediaSource &sender) override;
// 获取媒体源类型
mediakit::MediaOriginType getOriginType(mediakit::MediaSource &sender) const override;
MediaOriginType getOriginType(MediaSource &sender) const override;
// 获取媒体源url或者文件路径
std::string getOriginUrl(mediakit::MediaSource &sender) const override;
std::string getOriginUrl(MediaSource &sender) const override;
// 获取媒体源客户端相关信息
std::shared_ptr<SockInfo> getOriginSock(mediakit::MediaSource &sender) const override;
std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
// 获取丢包率
float getLossRate(mediakit::MediaSource &sender,mediakit::TrackType type) override;
float getLossRate(MediaSource &sender,TrackType type) override;
private:
WebRtcPusher(const EventPoller::Ptr &poller, const mediakit::RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership, const mediakit::MediaInfo &info, const mediakit::ProtocolOption &option);
WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
private:
bool _simulcast = false;
//断连续推延时
uint32_t _continue_push_ms = 0;
//媒体相关元数据
mediakit::MediaInfo _media_info;
MediaInfo _media_info;
//推流的rtsp源
mediakit::RtspMediaSourceImp::Ptr _push_src;
RtspMediaSourceImp::Ptr _push_src;
//推流所有权
std::shared_ptr<void> _push_src_ownership;
//推流的rtsp源,支持simulcast
std::unordered_map<std::string/*rid*/, mediakit::RtspMediaSource::Ptr> _push_src_sim;
std::unordered_map<std::string/*rid*/, RtspMediaSource::Ptr> _push_src_sim;
std::unordered_map<std::string/*rid*/, std::shared_ptr<void> > _push_src_sim_ownership;
};
}// namespace mediakit
#endif //ZLMEDIAKIT_WEBRTCPUSHER_H
webrtc/WebRtcSession.cpp
......@@ -12,7 +12,8 @@
#include "Util/util.h"
using namespace std;
using namespace mediakit;
namespace mediakit {
static string getUserName(const Buffer::Ptr &buffer) {
auto buf = buffer->data();
......@@ -85,7 +86,7 @@ void WebRtcSession::onError(const SockException &err) {
}
void WebRtcSession::onManager() {
GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
GET_CONFIG(float, timeoutSec, Rtc::kTimeOutSec);
if (!_transport && _ticker.createdTime() > timeoutSec * 1000) {
shutdown(SockException(Err_timeout, "illegal webrtc connection"));
return;
......@@ -96,5 +97,6 @@ void WebRtcSession::onManager() {
}
}
}// namespace mediakit
webrtc/WebRtcSession.h
......@@ -16,6 +16,8 @@
#include "IceServer.hpp"
#include "WebRtcTransport.h"
namespace mediakit {
class WebRtcSession : public UdpSession {
public:
WebRtcSession(const Socket::Ptr &sock);
......@@ -35,5 +37,6 @@ private:
std::shared_ptr<WebRtcTransportImp> _transport;
};
}// namespace mediakit
#endif //ZLMEDIAKIT_WEBRTCSESSION_H
webrtc/WebRtcTransport.cpp
......@@ -8,15 +8,18 @@
* may be found in the AUTHORS file in the root of the source tree.
*/
#include "WebRtcTransport.h"
#include <iostream>
#include <srtp2/srtp.h>
#include "RtpExt.h"
#include "Rtcp/Rtcp.h"
#include "Rtcp/RtcpFCI.h"
#include "RtpExt.h"
#include "Rtsp/RtpReceiver.h"
#include "WebRtcTransport.h"
#include <srtp2/srtp.h>
#include <iostream>
#include "WebRtcEchoTest.h"
#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
#define RTP_SSRC_OFFSET 1
#define RTX_SSRC_OFFSET 2
......@@ -26,10 +29,11 @@
#define RTP_MSID RTP_MSLABEL " " RTP_LABEL
using namespace std;
using namespace mediakit;
namespace mediakit {
// RTC配置项目
namespace RTC {
namespace Rtc {
#define RTC_FIELD "rtc."
// rtp和rtcp接受超时时间
const string kTimeOutSec = RTC_FIELD "timeoutSec";
......@@ -246,7 +250,7 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
// 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);
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
configure.enableTWCC(!remb_bit_rate);
}
......@@ -368,7 +372,7 @@ void WebRtcTransportImp::onCreate() {
registerSelf();
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
GET_CONFIG(float, timeoutSec, Rtc::kTimeOutSec);
_timer = std::make_shared<Timer>(
timeoutSec / 2,
[weak_self]() {
......@@ -511,7 +515,7 @@ void WebRtcTransportImp::onStartWebRTC() {
void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
// 修改answer sdp的ip、端口信息
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str) {
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, Rtc::kExternIP, [](string str) {
std::vector<std::string> ret;
if (str.length()) {
ret = split(str, ",");
......@@ -525,7 +529,7 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
GET_CONFIG(uint16_t, local_port, RTC::kPort);
GET_CONFIG(uint16_t, local_port, Rtc::kPort);
m.rtcp_addr.port = local_port;
m.port = m.rtcp_addr.port;
sdp.origin.address = m.addr.address;
......@@ -592,9 +596,9 @@ makeIceCandidate(std::string ip, uint16_t port, uint32_t priority = 100, std::st
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
GET_CONFIG(uint16_t, local_port, RTC::kPort);
GET_CONFIG(uint16_t, local_port, Rtc::kPort);
// 添加接收端口candidate信息
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str) {
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, Rtc::kExternIP, [](string str) {
std::vector<std::string> ret;
if (str.length()) {
ret = split(str, ",");
......@@ -698,7 +702,7 @@ std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const {
return it_chn->second;
}
float WebRtcTransportImp::getLossRate(mediakit::TrackType type) {
float WebRtcTransportImp::getLossRate(TrackType type) {
for (auto &pr : _ssrc_to_track) {
auto ssrc = pr.first;
auto &track = pr.second;
......@@ -942,7 +946,7 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPa
sendRtcpPli(rtp->getSSRC());
// 开启remb，则发送remb包调节比特率
GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
if (remb_bit_rate && _answer_sdp->supportRtcpFb(SdpConst::kRembRtcpFb)) {
sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
}
......@@ -1121,10 +1125,6 @@ void WebRtcPluginManager::getAnswerSdp(
it->second(sender, offer, args, cb);
}
#include "WebRtcEchoTest.h"
#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
void echo_plugin(
Session &sender, const string &offer, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
......@@ -1227,3 +1227,5 @@ static onceToken s_rtc_auto_register([]() {
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
});
}// namespace mediakit
\ No newline at end of file
webrtc/WebRtcTransport.h
......@@ -27,8 +27,10 @@
#include "TwccContext.h"
#include "SctpAssociation.hpp"
namespace mediakit {
//RTC配置项目
namespace RTC {
namespace Rtc {
extern const std::string kPort;
extern const std::string kTimeOutSec;
}//namespace RTC
......@@ -200,7 +202,7 @@ public:
//for send rtp
NackList nack_list;
mediakit::RtcpContext::Ptr rtcp_context_send;
RtcpContext::Ptr rtcp_context_send;
//for recv rtp
std::unordered_map<std::string/*rid*/, std::shared_ptr<RtpChannel> > rtp_channel;
......@@ -211,13 +213,13 @@ struct WrappedMediaTrack {
MediaTrack::Ptr track;
explicit WrappedMediaTrack(MediaTrack::Ptr ptr): track(ptr) {}
virtual ~WrappedMediaTrack() {}
virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, mediakit::RtpHeader *rtp) = 0;
virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) = 0;
};
struct WrappedRtxTrack: public WrappedMediaTrack {
explicit WrappedRtxTrack(MediaTrack::Ptr ptr)
: WrappedMediaTrack(std::move(ptr)) {}
void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, mediakit::RtpHeader *rtp) override;
void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
};
class WebRtcTransportImp;
......@@ -229,7 +231,7 @@ struct WrappedRtpTrack : public WrappedMediaTrack {
, _transport(t) {}
TwccContext& _twcc_ctx;
WebRtcTransportImp& _transport;
void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, mediakit::RtpHeader *rtp) override;
void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
};
class WebRtcTransportImp : public WebRtcTransport {
......@@ -243,7 +245,7 @@ public:
uint64_t getDuration() const;
bool canSendRtp() const;
bool canRecvRtp() const;
void onSendRtp(const mediakit::RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
......@@ -262,14 +264,14 @@ protected:
void onCreate() override;
void onDestory() override;
void onShutdown(const SockException &ex) override;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) = 0;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
void updateTicker();
float getLossRate(mediakit::TrackType type);
float getLossRate(TrackType type);
void onRtcpBye() override;
private:
void onSortedRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp);
void onSendNack(MediaTrack &track, const mediakit::FCI_NACK &nack, uint32_t ssrc);
void onSortedRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp);
void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
void onSendTwcc(uint32_t ssrc, const std::string &twcc_fci);
void registerSelf();
......@@ -343,4 +345,6 @@ private:
private:
mutable std::mutex _mtx_creator;
std::unordered_map<std::string, Plugin> _map_creator;
};
\ No newline at end of file
};
}// namespace mediakit
\ No newline at end of file
webrtc/readme.md
......@@ -15,7 +15,10 @@
- srtp相关功能：
- SrtpSession.cpp
- SrtpSession.hpp
- datachannel相关功能：
- SctpAssociation.cpp
- SctpAssociation.hpp
以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者，
用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。
......
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