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Cleanup and reformatting

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This commit is contained in:
Paul-Louis Ageneau
2020-11-28 16:37:46 +01:00
parent ffe202a6a2
commit 244c834992
9 changed files with 125 additions and 99 deletions
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32
include/rtc/description.hpp
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@@ -32,6 +32,15 @@
namespace rtc {
const string DEFAULT_AUDIO_PROFILE =
"minptime=10;maxaveragebitrate=96000;stereo=1;sprop-stereo=1;useinbandfec=1";
// Use Constrained Baseline profile Level 4.2 (necessary for Firefox)
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#Supported_video_codecs
// TODO: Should be 42E0 but 42C0 appears to be more compatible. Investigate this.
const string DEFAULT_VIDEO_PROFILE =
"profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1";
class RTC_CPP_EXPORT Description {
public:
enum class Type { Unspec, Offer, Answer, Pranswer, Rollback };
@@ -131,8 +140,10 @@ public:
void removeFormat(const string &fmt);
void addSSRC(uint32_t ssrc, std::optional<std::string> name, std::optional<std::string> msid=std::nullopt);
void replaceSSRC(uint32_t oldSSRC, uint32_t ssrc, std::optional<std::string> name, std::optional<std::string> msid=std::nullopt);
void addSSRC(uint32_t ssrc, std::optional<string> name,
std::optional<string> msid = nullopt);
void replaceSSRC(uint32_t oldSSRC, uint32_t ssrc, std::optional<string> name,
std::optional<string> msid = nullopt);
bool hasSSRC(uint32_t ssrc);
std::vector<uint32_t> getSSRCs();
@@ -141,7 +152,8 @@ public:
bool hasPayloadType(int payloadType) const;
void addRTXCodec(unsigned int payloadType, unsigned int originalPayloadType, unsigned int clockRate);
void addRTXCodec(unsigned int payloadType, unsigned int originalPayloadType,
unsigned int clockRate);
virtual void parseSdpLine(string_view line) override;
@@ -151,7 +163,7 @@ public:
void removeFB(const string &string);
void addFB(const string &string);
void addAttribute(std::string attr) { fmtps.emplace_back(attr); }
void addAttribute(string attr) { fmtps.emplace_back(std::move(attr)); }
int pt;
string format;
@@ -190,8 +202,9 @@ public:
public:
Audio(string mid = "audio", Direction dir = Direction::SendOnly);
void addAudioCodec(int payloadType, const string &codec, const std::optional<std::string>& profile=
"minptime=10; maxaveragebitrate=96000; stereo=1; sprop-stereo=1; useinbandfec=1");
void addAudioCodec(int payloadType, string codec,
std::optional<string> profile = DEFAULT_AUDIO_PROFILE);
void addOpusCodec(int payloadType);
};
@@ -199,11 +212,8 @@ public:
public:
Video(string mid = "video", Direction dir = Direction::SendOnly);
// Use Constrained Baseline profile Level 4.2 (necessary for Firefox)
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#Supported_video_codecs
// TODO: Should be 42E0 but 42C0 appears to be more compatible. Investigate this.
void addVideoCodec(int payloadType, const string &codec, const std::optional<std::string>&
profile="profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1");
void addVideoCodec(int payloadType, string codec,
std::optional<string> profile = DEFAULT_VIDEO_PROFILE);
void addH264Codec(int payloadType);
void addVP8Codec(int payloadType);

44
include/rtc/rtp.hpp
View File

@@ -66,10 +66,17 @@ public:
inline uint32_t ssrc() const { return ntohl(_ssrc); }
inline size_t getSize() const {
return ((char *)&csrc) - ((char *)this) + sizeof(SSRC) * csrcCount();
return reinterpret_cast<const char *>(&csrc) - reinterpret_cast<const char *>(this) +
sizeof(SSRC) * csrcCount();
}
char *getBody() const { return ((char *)&csrc) + sizeof(SSRC) * csrcCount(); }
[[nodiscard]] char *getBody() {
return reinterpret_cast<char *>(&csrc) + sizeof(SSRC) * csrcCount();
}
[[nodiscard]] const char *getBody() const {
return reinterpret_cast<const char *>(&csrc) + sizeof(SSRC) * csrcCount();
}
inline void setSeqNumber(uint16_t newSeqNo) { _seqNumber = htons(newSeqNo); }
inline void setPayloadType(uint8_t newPayloadType) {
@@ -117,18 +124,19 @@ public:
}
inline void setSSRC(SSRC in_ssrc) { this->ssrc = htonl(in_ssrc); }
inline SSRC getSSRC() const { return ntohl(ssrc); }
[[nodiscard]] inline SSRC getSSRC() const { return ntohl(ssrc); }
inline void setPacketsLost([[maybe_unused]] unsigned int packetsLost,
[[maybe_unused]] unsigned int totalPackets) {
// TODO Implement loss percentages.
_fractionLostAndPacketsLost = 0;
}
inline unsigned int getLossPercentage() const {
[[nodiscard]] inline unsigned int getLossPercentage() const {
// TODO Implement loss percentages.
return 0;
}
inline unsigned int getPacketLostCount() const {
[[nodiscard]] inline unsigned int getPacketLostCount() const {
// TODO Implement total packets lost.
return 0;
}
@@ -145,13 +153,13 @@ public:
inline void setJitter(uint32_t jitter) { _jitter = htonl(jitter); }
inline void setNTPOfSR(uint64_t ntp) { _lastReport = htonll(ntp >> 16u); }
inline uint32_t getNTPOfSR() const { return ntohl(_lastReport) << 16u; }
[[nodiscard]] inline uint32_t getNTPOfSR() const { return ntohl(_lastReport) << 16u; }
inline void setDelaySinceSR(uint32_t sr) {
// The delay, expressed in units of 1/65536 seconds
_delaySinceLastReport = htonl(sr);
}
inline uint32_t getDelaySinceSR() const { return ntohl(_delaySinceLastReport); }
[[nodiscard]] inline uint32_t getDelaySinceSR() const { return ntohl(_delaySinceLastReport); }
inline void log() const {
PLOG_VERBOSE << "RTCP report block: "
@@ -242,8 +250,10 @@ public:
this->_senderSSRC = htonl(senderSSRC);
}
inline RTCP_ReportBlock *getReportBlock(int num) { return &_reportBlocks + num; }
inline const RTCP_ReportBlock *getReportBlock(int num) const { return &_reportBlocks + num; }
[[nodiscard]] inline RTCP_ReportBlock *getReportBlock(int num) { return &_reportBlocks + num; }
[[nodiscard]] inline const RTCP_ReportBlock *getReportBlock(int num) const {
return &_reportBlocks + num;
}
[[nodiscard]] inline size_t getSize() const {
// "length" in packet is one less than the number of 32 bit words in the packet.
@@ -280,8 +290,10 @@ private:
RTCP_ReportBlock _reportBlocks;
public:
inline RTCP_ReportBlock *getReportBlock(int num) { return &_reportBlocks + num; }
inline const RTCP_ReportBlock *getReportBlock(int num) const { return &_reportBlocks + num; }
[[nodiscard]] inline RTCP_ReportBlock *getReportBlock(int num) { return &_reportBlocks + num; }
[[nodiscard]] inline const RTCP_ReportBlock *getReportBlock(int num) const {
return &_reportBlocks + num;
}
inline SSRC senderSSRC() const { return ntohl(_senderSSRC); }
inline void setSenderSSRC(SSRC ssrc) { this->_senderSSRC = htonl(ssrc); }
@@ -477,11 +489,15 @@ public:
return ntohs(*(uint16_t *)(header.getBody()));
}
char *getBody() { return header.getBody() + sizeof(uint16_t); }
[[nodiscard]] char *getBody() { return header.getBody() + sizeof(uint16_t); }
size_t getBodySize(size_t totalSize) { return totalSize - ((char *)getBody() - (char *)this); }
[[nodiscard]] const char *getBody() const { return header.getBody() + sizeof(uint16_t); }
RTP &getHeader() { return header; }
[[nodiscard]] size_t getBodySize(size_t totalSize) {
return totalSize - (getBody() - reinterpret_cast<char *>(this));
}
[[nodiscard]] RTP &getHeader() { return header; }
size_t normalizePacket(size_t totalSize, SSRC originalSSRC, uint8_t originalPayloadType) {
header.setSeqNumber(getOriginalSeqNo());

2
include/rtc/websocket.hpp
View File

@@ -49,7 +49,7 @@ public:
struct Configuration {
bool disableTlsVerification = false; // if true, don't verify the TLS certificate
std::optional<std::vector<string>> protocols = std::nullopt;
std::vector<string> protocols;
};
WebSocket(std::optional<Configuration> config = nullopt);

36
src/description.cpp
View File

@@ -29,13 +29,13 @@
#include <unordered_map>
using std::shared_ptr;
using std::size_t;
using std::string;
using std::string_view;
using std::chrono::system_clock;
namespace {
using std::string;
using std::string_view;
inline bool match_prefix(string_view str, string_view prefix) {
return str.size() >= prefix.size() &&
std::mismatch(prefix.begin(), prefix.end(), str.begin()).first == prefix.end();
@@ -483,8 +483,7 @@ string Description::Entry::generateSdpLines(string_view eol) const {
}
for (const auto &attr : mAttributes) {
if (attr.find("extmap") == std::string::npos &&
attr.find("rtcp-rsize") == std::string::npos)
if (attr.find("extmap") == string::npos && attr.find("rtcp-rsize") == string::npos)
sdp << "a=" << attr << eol;
}
@@ -519,7 +518,8 @@ Description::Entry::removeAttribute(std::vector<string>::iterator it) {
return mAttributes.erase(it);
}
void Description::Media::addSSRC(uint32_t ssrc, std::optional<std::string> name, std::optional<std::string> msid) {
void Description::Media::addSSRC(uint32_t ssrc, std::optional<string> name,
std::optional<string> msid) {
if (name)
mAttributes.emplace_back("ssrc:" + std::to_string(ssrc) + " cname:" + *name);
else
@@ -531,7 +531,8 @@ void Description::Media::addSSRC(uint32_t ssrc, std::optional<std::string> name,
mSsrcs.emplace_back(ssrc);
}
void Description::Media::replaceSSRC(uint32_t oldSSRC, uint32_t ssrc, std::optional<std::string> name, std::optional<std::string> msid) {
void Description::Media::replaceSSRC(uint32_t oldSSRC, uint32_t ssrc, std::optional<string> name,
std::optional<string> msid) {
auto it = mAttributes.begin();
while (it != mAttributes.end()) {
if (it->find("ssrc:" + std::to_string(oldSSRC)) == 0) {
@@ -702,7 +703,8 @@ void Description::Media::removeFormat(const string &fmt) {
}
}
void Description::Video::addVideoCodec(int payloadType, const string &codec, const std::optional<std::string>& profile) {
void Description::Video::addVideoCodec(int payloadType, string codec,
std::optional<string> profile) {
RTPMap map(std::to_string(payloadType) + ' ' + codec + "/90000");
map.addFB("nack");
map.addFB("nack pli");
@@ -712,7 +714,6 @@ void Description::Video::addVideoCodec(int payloadType, const string &codec, con
map.fmtps.emplace_back(*profile);
addRTPMap(map);
// // RTX Packets
/* TODO
* TIL that Firefox does not properly support the negotiation of RTX! It works, but doesn't
* negotiate the SSRC so we have no idea what SSRC is RTX going to be. Three solutions: One) we
@@ -720,13 +721,15 @@ void Description::Video::addVideoCodec(int payloadType, const string &codec, con
* rebuild the original packet before we send it distribute it to each track. Three) we complain
* to mozilla. This one probably won't do much.
*/
// RTX Packets
// RTPMap rtx(std::to_string(payloadType+1) + " rtx/90000");
// // TODO rtx-time is how long can a request be stashed for before needing to resend it.
// Needs to be parameterized rtx.addAttribute("apt=" + std::to_string(payloadType) +
// ";rtx-time=3000"); addRTPMap(rtx);
}
void Description::Audio::addAudioCodec(int payloadType, const string &codec, const std::optional<std::string>& profile) {
void Description::Audio::addAudioCodec(int payloadType, string codec,
std::optional<string> profile) {
// TODO This 48000/2 should be parameterized
RTPMap map(std::to_string(payloadType) + ' ' + codec + "/48000/2");
if (profile)
@@ -734,7 +737,8 @@ void Description::Audio::addAudioCodec(int payloadType, const string &codec, con
addRTPMap(map);
}
void Description::Media::addRTXCodec(unsigned int payloadType, unsigned int originalPayloadType, unsigned int clockRate) {
void Description::Media::addRTXCodec(unsigned int payloadType, unsigned int originalPayloadType,
unsigned int clockRate) {
RTPMap map(std::to_string(payloadType) + " RTX/" + std::to_string(clockRate));
map.fmtps.emplace_back("apt=" + std::to_string(originalPayloadType));
addRTPMap(map);
@@ -742,9 +746,9 @@ void Description::Media::addRTXCodec(unsigned int payloadType, unsigned int orig
void Description::Video::addH264Codec(int pt) { addVideoCodec(pt, "H264"); }
void Description::Video::addVP8Codec(int payloadType) { addVideoCodec(payloadType, "VP8"); }
void Description::Video::addVP8Codec(int payloadType) { addVideoCodec(payloadType, "VP8", nullopt); }
void Description::Video::addVP9Codec(int payloadType) { addVideoCodec(payloadType, "VP9"); }
void Description::Video::addVP9Codec(int payloadType) { addVideoCodec(payloadType, "VP9", nullopt); }
void Description::Media::setBitrate(int bitrate) { mBas = bitrate; }
@@ -813,7 +817,7 @@ void Description::Media::parseSdpLine(string_view line) {
} else if (key == "rtcp-mux") {
// always added
} else if (key == "ssrc") {
mSsrcs.emplace_back(std::stoul((std::string)value));
mSsrcs.emplace_back(std::stoul(string(value)));
} else {
Entry::parseSdpLine(line);
}
@@ -833,7 +837,7 @@ std::vector<uint32_t> Description::Media::getSSRCs() {
for (auto &val : mAttributes) {
PLOG_DEBUG << val;
if (val.find("ssrc:") == 0) {
vec.emplace_back(std::stoul((std::string)val.substr(5, val.find(" "))));
vec.emplace_back(std::stoul(string(val.substr(5, val.find(" ")))));
}
}
return vec;
@@ -857,7 +861,7 @@ Description::Media::RTPMap::RTPMap(string_view mline) { setMLine(mline); }
void Description::Media::RTPMap::removeFB(const string &str) {
auto it = rtcpFbs.begin();
while (it != rtcpFbs.end()) {
if (it->find(str) != std::string::npos) {
if (it->find(str) != string::npos) {
it = rtcpFbs.erase(it);
} else
it++;

16
src/dtlssrtptransport.cpp
View File

@@ -110,7 +110,7 @@ bool DtlsSrtpTransport::sendMedia(message_ptr message) {
if (err == srtp_err_status_replay_fail)
throw std::runtime_error("SRTCP packet is a replay");
else if (err == srtp_err_status_no_ctx) {
auto ssrc = ((RTCP_SR *)message->data())->senderSSRC();
auto ssrc = reinterpret_cast<RTCP_SR *>(message->data())->senderSSRC();
PLOG_INFO << "Adding SSRC to SRTCP: " << ssrc;
addSSRC(ssrc);
if ((err = srtp_protect_rtcp(mSrtpOut, message->data(), &size)))
@@ -127,7 +127,7 @@ bool DtlsSrtpTransport::sendMedia(message_ptr message) {
if (err == srtp_err_status_replay_fail)
throw std::runtime_error("Outgoing SRTP packet is a replay");
else if (err == srtp_err_status_no_ctx) {
auto ssrc = ((RTP *)message->data())->ssrc();
auto ssrc = reinterpret_cast<RTP *>(message->data())->ssrc();
PLOG_INFO << "Adding SSRC to RTP: " << ssrc;
addSSRC(ssrc);
if ((err = srtp_protect(mSrtpOut, message->data(), &size)))
@@ -196,7 +196,7 @@ void DtlsSrtpTransport::incoming(message_ptr message) {
else if (err == srtp_err_status_auth_fail)
PLOG_WARNING << "Incoming SRTCP packet failed authentication check";
else if (err == srtp_err_status_no_ctx) {
auto ssrc = ((RTCP_SR *)message->data())->senderSSRC();
auto ssrc = reinterpret_cast<RTCP_SR *>(message->data())->senderSSRC();
PLOG_INFO << "Adding SSRC to RTCP: " << ssrc;
addSSRC(ssrc);
if ((err = srtp_unprotect_rtcp(mSrtpIn, message->data(), &size)))
@@ -210,8 +210,7 @@ void DtlsSrtpTransport::incoming(message_ptr message) {
}
PLOG_VERBOSE << "Unprotected SRTCP packet, size=" << size;
message->type = Message::Type::Control;
auto rtp = (RTCP_SR *)message->data();
message->stream = rtp->senderSSRC();
message->stream = reinterpret_cast<RTCP_SR *>(message->data())->senderSSRC();
} else {
PLOG_VERBOSE << "Incoming SRTP packet, size=" << size;
if (srtp_err_status_t err = srtp_unprotect(mSrtpIn, message->data(), &size)) {
@@ -220,7 +219,7 @@ void DtlsSrtpTransport::incoming(message_ptr message) {
else if (err == srtp_err_status_auth_fail)
PLOG_WARNING << "Incoming SRTP packet failed authentication check";
else if (err == srtp_err_status_no_ctx) {
auto ssrc = ((RTP *)message->data())->ssrc();
auto ssrc = reinterpret_cast<RTP *>(message->data())->ssrc();
PLOG_INFO << "Adding SSRC to RTP: " << ssrc;
addSSRC(ssrc);
if ((err = srtp_unprotect(mSrtpIn, message->data(), &size)))
@@ -228,13 +227,12 @@ void DtlsSrtpTransport::incoming(message_ptr message) {
to_string(static_cast<int>(err)));
} else
PLOG_WARNING << "SRTP unprotect error, status=" << err
<< " SSRC=" << ((RTP *)message->data())->ssrc();
<< " SSRC=" << reinterpret_cast<RTP *>(message->data())->ssrc();
return;
}
PLOG_VERBOSE << "Unprotected SRTP packet, size=" << size;
message->type = Message::Type::Binary;
auto rtp = (RTP *)message->data();
message->stream = rtp->ssrc();
message->stream = reinterpret_cast<RTP *>(message->data())->ssrc();
}
message->resize(size);

2
src/rtcp.cpp
View File

@@ -126,7 +126,7 @@ bool RtcpReceivingSession::requestKeyframe() {
void RtcpReceivingSession::pushPLI() {
auto msg = rtc::make_message(rtc::RTCP_PLI::size(), rtc::Message::Type::Control);
auto *pli = (rtc::RTCP_PLI *)msg->data();
auto *pli = reinterpret_cast<rtc::RTCP_PLI *>(msg->data());
pli->preparePacket(mSsrc);
send(msg);
}

11
src/websocket.cpp
View File

@@ -292,14 +292,13 @@ shared_ptr<WsTransport> WebSocket::initWsTransport() {
if (!lower)
lower = std::atomic_load(&mTcpTransport);
auto wsConfig = WsTransport::Configuration();
if(mConfig.protocols) {
wsConfig.protocols = *mConfig.protocols;
}
WsTransport::Configuration wsConfig = {};
wsConfig.host = mHost;
wsConfig.path = mPath;
wsConfig.protocols = mConfig.protocols;
auto transport = std::make_shared<WsTransport>(
wsConfig,
lower, mHost, mPath, weak_bind(&WebSocket::incoming, this, _1),
lower, wsConfig, weak_bind(&WebSocket::incoming, this, _1),
[this, weak_this = weak_from_this()](State state) {
auto shared_this = weak_this.lock();
if (!shared_this)

27
src/wstransport.cpp
View File

@@ -27,9 +27,9 @@
#include <iterator>
#include <list>
#include <map>
#include <numeric>
#include <random>
#include <regex>
#include <numeric>
#ifdef _WIN32
#include <winsock2.h>
@@ -54,17 +54,17 @@ using std::to_string;
using random_bytes_engine =
std::independent_bits_engine<std::default_random_engine, CHAR_BIT, unsigned short>;
WsTransport::WsTransport(std::optional<Configuration> config, std::shared_ptr<Transport> lower, string host, string path,
WsTransport::WsTransport(std::shared_ptr<Transport> lower, Configuration config,
message_callback recvCallback, state_callback stateCallback)
: Transport(lower, std::move(stateCallback)), mHost(std::move(host)), mPath(std::move(path)), mConfig(config ? std::move(*config) : Configuration()) {
: Transport(lower, std::move(stateCallback)), mConfig(std::move(config)) {
onRecv(recvCallback);
PLOG_DEBUG << "Initializing WebSocket transport";
if (mHost.empty())
if (mConfig.host.empty())
throw std::invalid_argument("WebSocket HTTP host cannot be empty");
if (mPath.empty())
if (mConfig.path.empty())
throw std::invalid_argument("WebSocket HTTP path cannot be empty");
}
@@ -155,7 +155,7 @@ void WsTransport::close() {
}
bool WsTransport::sendHttpRequest() {
PLOG_DEBUG << "Sending WebSocket HTTP request for path " << mPath;
PLOG_DEBUG << "Sending WebSocket HTTP request for path " << mConfig.path;
changeState(State::Connecting);
auto seed = static_cast<unsigned int>(system_clock::now().time_since_epoch().count());
@@ -167,26 +167,25 @@ bool WsTransport::sendHttpRequest() {
string appendHeader = "";
if (mConfig.protocols.size() > 0) {
appendHeader += "Sec-WebSocket-Protocol: " +
std::accumulate(mConfig.protocols.begin(), mConfig.protocols.end(), string(), [](const string& a, const string& b) -> string {
appendHeader +=
"Sec-WebSocket-Protocol: " +
std::accumulate(mConfig.protocols.begin(), mConfig.protocols.end(), string(),
[](const string &a, const string &b) -> string {
return a + (a.length() > 0 ? "," : "") + b;
}) +
"\r\n";
}
const string request = "GET " + mPath +
const string request = "GET " + mConfig.path +
" HTTP/1.1\r\n"
"Host: " +
mHost +
mConfig.host +
"\r\n"
"Connection: Upgrade\r\n"
"Upgrade: websocket\r\n"
"Sec-WebSocket-Version: 13\r\n"
"Sec-WebSocket-Key: " +
to_base64(key) +
"\r\n" +
std::move(appendHeader) +
"\r\n";
to_base64(key) + "\r\n" + std::move(appendHeader) + "\r\n";
auto data = reinterpret_cast<const byte *>(request.data());
auto size = request.size();

8
src/wstransport.hpp
View File

@@ -32,10 +32,12 @@ class TlsTransport;
class WsTransport : public Transport {
public:
struct Configuration {
string host;
string path = "/";
std::vector<string> protocols;
};
WsTransport(std::optional<Configuration> config, std::shared_ptr<Transport> lower, string host, string path,
WsTransport(std::shared_ptr<Transport> lower, Configuration config,
message_callback recvCallback, state_callback stateCallback);
~WsTransport();
@@ -72,14 +74,12 @@ private:
void recvFrame(const Frame &frame);
bool sendFrame(const Frame &frame);
const string mHost;
const string mPath;
const Configuration mConfig;
binary mBuffer;
binary mPartial;
Opcode mPartialOpcode;
const Configuration mConfig;
};
} // namespace rtc