| 123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208209210211212213214215216217218219220221222223224225226227228229230231232233234235236237238239240241242243244245246247248249250251252253254255256257258259260261262263264265266267268269270271272273274275276277278279280281282283284285286287288289290291292293294295296297298299300301302303304305306307308309310311312313314315316317318319320321322323324325326327328329330331332333334335336337338339340341342343344345346347348349350351352353354355356357358359360361362363364365366367368369370371372373374375376377378379380381382383384385386387388389390391392393394395396397398399400401402403404405406407408409410411412413414415416417418419420421422423424425426427428429430431432433434435436437438439440441442443444445446447448 | /** * libdatachannel client example * Copyright (c) 2019-2020 Paul-Louis Ageneau * Copyright (c) 2019 Murat Dogan * Copyright (c) 2020 Will Munn * Copyright (c) 2020 Nico Chatzi * Copyright (c) 2020 Lara Mackey * Copyright (c) 2020 Erik Cota-Robles * Copyright (c) 2020 Filip Klembara (in2core) * * This Source Code Form is subject to the terms of the Mozilla Public * License, v. 2.0. If a copy of the MPL was not distributed with this * file, You can obtain one at https://mozilla.org/MPL/2.0/. */#include "nlohmann/json.hpp"#include "h264fileparser.hpp"#include "opusfileparser.hpp"#include "helpers.hpp"#include "ArgParser.hpp"using namespace rtc;using namespace std;using namespace std::chrono_literals;using json = nlohmann::json;template <class T> weak_ptr<T> make_weak_ptr(shared_ptr<T> ptr) { return ptr; }/// all connected clientsunordered_map<string, shared_ptr<Client>> clients{};/// Creates peer connection and client representation/// @param config Configuration/// @param wws Websocket for signaling/// @param id Client ID/// @returns Clientshared_ptr<Client> createPeerConnection(const Configuration &config,                                        weak_ptr<WebSocket> wws,                                        string id);/// Creates stream/// @param h264Samples Directory with H264 samples/// @param fps Video FPS/// @param opusSamples Directory with opus samples/// @returns Stream objectshared_ptr<Stream> createStream(const string h264Samples, const unsigned fps, const string opusSamples);/// Add client to stream/// @param client Client/// @param adding_video True if adding videovoid addToStream(shared_ptr<Client> client, bool isAddingVideo);/// Start streamvoid startStream();/// Main dispatch queueDispatchQueue MainThread("Main");/// Audio and video streamoptional<shared_ptr<Stream>> avStream = nullopt;const string defaultRootDirectory = "../../../examples/streamer/samples/";const string defaultH264SamplesDirectory = defaultRootDirectory + "h264/";string h264SamplesDirectory = defaultH264SamplesDirectory;const string defaultOpusSamplesDirectory = defaultRootDirectory + "opus/";string opusSamplesDirectory = defaultOpusSamplesDirectory;const string defaultIPAddress = "127.0.0.1";const uint16_t defaultPort = 8000;string ip_address = defaultIPAddress;uint16_t port = defaultPort;/// Incomming message handler for websocket/// @param message Incommint message/// @param config Configuration/// @param ws Websocketvoid wsOnMessage(json message, Configuration config, shared_ptr<WebSocket> ws) {    auto it = message.find("id");    if (it == message.end())        return;    string id = it->get<string>();    it = message.find("type");    if (it == message.end())        return;    string type = it->get<string>();    if (type == "request") {        clients.emplace(id, createPeerConnection(config, make_weak_ptr(ws), id));    } else if (type == "answer") {        if (auto jt = clients.find(id); jt != clients.end()) {            auto pc = jt->second->peerConnection;            auto sdp = message["sdp"].get<string>();            auto description = Description(sdp, type);            pc->setRemoteDescription(description);        }    }}int main(int argc, char **argv) try {    bool enableDebugLogs = false;    bool printHelp = false;    int c = 0;    auto parser = ArgParser({{"a", "audio"}, {"b", "video"}, {"d", "ip"}, {"p","port"}}, {{"h", "help"}, {"v", "verbose"}});    auto parsingResult = parser.parse(argc, argv, [](string key, string value) {        if (key == "audio") {            opusSamplesDirectory = value + "/";        } else if (key == "video") {            h264SamplesDirectory = value + "/";        } else if (key == "ip") {            ip_address = value;        } else if (key == "port") {            port = atoi(value.data());        } else {            cerr << "Invalid option --" << key << " with value " << value << endl;            return false;        }        return true;    }, [&enableDebugLogs, &printHelp](string flag){        if (flag == "verbose") {            enableDebugLogs = true;        } else if (flag == "help") {            printHelp = true;        } else {            cerr << "Invalid flag --" << flag << endl;            return false;        }        return true;    });    if (!parsingResult) {        return 1;    }    if (printHelp) {        cout << "usage: stream-h264 [-a opus_samples_folder] [-b h264_samples_folder] [-d ip_address] [-p port] [-v] [-h]" << endl        << "Arguments:" << endl        << "\t -a " << "Directory with opus samples (default: " << defaultOpusSamplesDirectory << ")." << endl        << "\t -b " << "Directory with H264 samples (default: " << defaultH264SamplesDirectory << ")." << endl        << "\t -d " << "Signaling server IP address (default: " << defaultIPAddress << ")." << endl        << "\t -p " << "Signaling server port (default: " << defaultPort << ")." << endl        << "\t -v " << "Enable debug logs." << endl        << "\t -h " << "Print this help and exit." << endl;        return 0;    }    if (enableDebugLogs) {        InitLogger(LogLevel::Debug);    }    Configuration config;    string stunServer = "stun:stun.l.google.com:19302";    cout << "STUN server is " << stunServer << endl;    config.iceServers.emplace_back(stunServer);    config.disableAutoNegotiation = true;    string localId = "server";    cout << "The local ID is: " << localId << endl;    auto ws = make_shared<WebSocket>();    ws->onOpen([]() { cout << "WebSocket connected, signaling ready" << endl; });    ws->onClosed([]() { cout << "WebSocket closed" << endl; });    ws->onError([](const string &error) { cout << "WebSocket failed: " << error << endl; });    ws->onMessage([&](variant<binary, string> data) {        if (!holds_alternative<string>(data))            return;        json message = json::parse(get<string>(data));        MainThread.dispatch([message, config, ws]() {            wsOnMessage(message, config, ws);        });    });    const string url = "ws://" + ip_address + ":" + to_string(port) + "/" + localId;    cout << "URL is " << url << endl;    ws->open(url);    cout << "Waiting for signaling to be connected..." << endl;    while (!ws->isOpen()) {        if (ws->isClosed())            return 1;        this_thread::sleep_for(100ms);    }    while (true) {        string id;        cout << "Enter to exit" << endl;        cin >> id;        cin.ignore();        cout << "exiting" << endl;        break;    }    cout << "Cleaning up..." << endl;    return 0;} catch (const std::exception &e) {    std::cout << "Error: " << e.what() << std::endl;    return -1;}shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const uint8_t payloadType, const uint32_t ssrc, const string cname, const string msid, const function<void (void)> onOpen) {    auto video = Description::Video(cname);    video.addH264Codec(payloadType);    video.addSSRC(ssrc, cname, msid, cname);    auto track = pc->addTrack(video);    // create RTP configuration    auto rtpConfig = make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate);    // create packetizer    auto packetizer = make_shared<H264RtpPacketizer>(NalUnit::Separator::Length, rtpConfig);    // add RTCP SR handler    auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);    packetizer->addToChain(srReporter);    // add RTCP NACK handler    auto nackResponder = make_shared<RtcpNackResponder>();    packetizer->addToChain(nackResponder);    // set handler    track->setMediaHandler(packetizer);    track->onOpen(onOpen);    auto trackData = make_shared<ClientTrackData>(track, srReporter);    return trackData;}shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const uint8_t payloadType, const uint32_t ssrc, const string cname, const string msid, const function<void (void)> onOpen) {    auto audio = Description::Audio(cname);    audio.addOpusCodec(payloadType);    audio.addSSRC(ssrc, cname, msid, cname);    auto track = pc->addTrack(audio);    // create RTP configuration    auto rtpConfig = make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, OpusRtpPacketizer::DefaultClockRate);    // create packetizer    auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);    // add RTCP SR handler    auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);    packetizer->addToChain(srReporter);    // add RTCP NACK handler    auto nackResponder = make_shared<RtcpNackResponder>();    packetizer->addToChain(nackResponder);    // set handler    track->setMediaHandler(packetizer);    track->onOpen(onOpen);    auto trackData = make_shared<ClientTrackData>(track, srReporter);    return trackData;}// Create and setup a PeerConnectionshared_ptr<Client> createPeerConnection(const Configuration &config,                                                weak_ptr<WebSocket> wws,                                                string id) {    auto pc = make_shared<PeerConnection>(config);    auto client = make_shared<Client>(pc);    pc->onStateChange([id](PeerConnection::State state) {        cout << "State: " << state << endl;        if (state == PeerConnection::State::Disconnected ||            state == PeerConnection::State::Failed ||            state == PeerConnection::State::Closed) {            // remove disconnected client            MainThread.dispatch([id]() {                clients.erase(id);            });        }    });    pc->onGatheringStateChange(        [wpc = make_weak_ptr(pc), id, wws](PeerConnection::GatheringState state) {        cout << "Gathering State: " << state << endl;        if (state == PeerConnection::GatheringState::Complete) {            if(auto pc = wpc.lock()) {                auto description = pc->localDescription();                json message = {                    {"id", id},                    {"type", description->typeString()},                    {"sdp", string(description.value())}                };                // Gathering complete, send answer                if (auto ws = wws.lock()) {                    ws->send(message.dump());                }            }        }    });    client->video = addVideo(pc, 102, 1, "video-stream", "stream1", [id, wc = make_weak_ptr(client)]() {        MainThread.dispatch([wc]() {            if (auto c = wc.lock()) {                addToStream(c, true);            }        });        cout << "Video from " << id << " opened" << endl;    });    client->audio = addAudio(pc, 111, 2, "audio-stream", "stream1", [id, wc = make_weak_ptr(client)]() {        MainThread.dispatch([wc]() {            if (auto c = wc.lock()) {                addToStream(c, false);            }        });        cout << "Audio from " << id << " opened" << endl;    });    auto dc = pc->createDataChannel("ping-pong");    dc->onOpen([id, wdc = make_weak_ptr(dc)]() {        if (auto dc = wdc.lock()) {            dc->send("Ping");        }    });    dc->onMessage(nullptr, [id, wdc = make_weak_ptr(dc)](string msg) {        cout << "Message from " << id << " received: " << msg << endl;        if (auto dc = wdc.lock()) {            dc->send("Ping");        }    });    client->dataChannel = dc;    pc->setLocalDescription();    return client;};/// Create streamshared_ptr<Stream> createStream(const string h264Samples, const unsigned fps, const string opusSamples) {    // video source    auto video = make_shared<H264FileParser>(h264Samples, fps, true);    // audio source    auto audio = make_shared<OPUSFileParser>(opusSamples, true);    auto stream = make_shared<Stream>(video, audio);    // set callback responsible for sample sending    stream->onSample([ws = make_weak_ptr(stream)](Stream::StreamSourceType type, uint64_t sampleTime, rtc::binary sample) {        vector<ClientTrack> tracks{};        string streamType = type == Stream::StreamSourceType::Video ? "video" : "audio";        // get track for given type        function<optional<shared_ptr<ClientTrackData>> (shared_ptr<Client>)> getTrackData = [type](shared_ptr<Client> client) {            return type == Stream::StreamSourceType::Video ? client->video : client->audio;        };        // get all clients with Ready state        for(auto id_client: clients) {            auto id = id_client.first;            auto client = id_client.second;            auto optTrackData = getTrackData(client);            if (client->getState() == Client::State::Ready && optTrackData.has_value()) {                auto trackData = optTrackData.value();                tracks.push_back(ClientTrack(id, trackData));            }        }        if (!tracks.empty()) {            for (auto clientTrack: tracks) {                auto client = clientTrack.id;                auto trackData = clientTrack.trackData;                auto rtpConfig = trackData->sender->rtpConfig;                // sample time is in us, we need to convert it to seconds                auto elapsedSeconds = double(sampleTime) / (1000 * 1000);                // get elapsed time in clock rate                uint32_t elapsedTimestamp = rtpConfig->secondsToTimestamp(elapsedSeconds);                // set new timestamp                rtpConfig->timestamp = rtpConfig->startTimestamp + elapsedTimestamp;                // get elapsed time in clock rate from last RTCP sender report                auto reportElapsedTimestamp = rtpConfig->timestamp - trackData->sender->lastReportedTimestamp();                // check if last report was at least 1 second ago                if (rtpConfig->timestampToSeconds(reportElapsedTimestamp) > 1) {                    trackData->sender->setNeedsToReport();                }                cout << "Sending " << streamType << " sample with size: " << to_string(sample.size()) << " to " << client << endl;                try {                    // send sample                    trackData->track->send(sample);                } catch (const std::exception &e) {                    cerr << "Unable to send "<< streamType << " packet: " << e.what() << endl;                }            }        }        MainThread.dispatch([ws]() {            if (clients.empty()) {                // we have no clients, stop the stream                if (auto stream = ws.lock()) {                    stream->stop();                }            }        });    });    return stream;}/// Start streamvoid startStream() {    shared_ptr<Stream> stream;    if (avStream.has_value()) {        stream = avStream.value();        if (stream->isRunning) {            // stream is already running            return;        }    } else {        stream = createStream(h264SamplesDirectory, 30, opusSamplesDirectory);        avStream = stream;    }    stream->start();}/// Send previous key frame so browser can show something to user/// @param stream Stream/// @param video Video track datavoid sendInitialNalus(shared_ptr<Stream> stream, shared_ptr<ClientTrackData> video) {    auto h264 = dynamic_cast<H264FileParser *>(stream->video.get());    auto initialNalus = h264->initialNALUS();    // send previous NALU key frame so users don't have to wait to see stream works    if (!initialNalus.empty()) {        const double frameDuration_s = double(h264->getSampleDuration_us()) / (1000 * 1000);        const uint32_t frameTimestampDuration = video->sender->rtpConfig->secondsToTimestamp(frameDuration_s);        video->sender->rtpConfig->timestamp = video->sender->rtpConfig->startTimestamp - frameTimestampDuration * 2;        video->track->send(initialNalus);        video->sender->rtpConfig->timestamp += frameTimestampDuration;        // Send initial NAL units again to start stream in firefox browser        video->track->send(initialNalus);    }}/// Add client to stream/// @param client Client/// @param adding_video True if adding videovoid addToStream(shared_ptr<Client> client, bool isAddingVideo) {    if (client->getState() == Client::State::Waiting) {        client->setState(isAddingVideo ? Client::State::WaitingForAudio : Client::State::WaitingForVideo);    } else if ((client->getState() == Client::State::WaitingForAudio && !isAddingVideo)               || (client->getState() == Client::State::WaitingForVideo && isAddingVideo)) {        // Audio and video tracks are collected now        assert(client->video.has_value() && client->audio.has_value());        auto video = client->video.value();        if (avStream.has_value()) {            sendInitialNalus(avStream.value(), video);        }        client->setState(Client::State::Ready);    }    if (client->getState() == Client::State::Ready) {        startStream();    }}
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