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Thomas
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add h265 example
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‎examples/libdatachannel/CMakeLists.txt‎

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@@ -20,3 +20,4 @@ FetchContent_MakeAvailable(NabtoWebrtcSignaling)
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add_subdirectory(src/common)
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add_subdirectory(src/webrtc_device)
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add_subdirectory(src/webrtc_device_rtsp)
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add_subdirectory(src/webrtc_device_rtsp_h265)

‎examples/libdatachannel/src/webrtc_device_rtsp_h265/CMakeLists.txt‎

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set(src
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main.cpp
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rtsp-client/rtsp_client.cpp
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rtsp-client/tcp_rtp_client.cpp
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)
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option(RTSP_HAS_BASIC_AUTH "disable RTSP client basic auth" ON)
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option(RTSP_HAS_DIGEST_AUTH "disable RTSP client digest auth" ON)
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if (RTSP_HAS_BASIC_AUTH)
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add_definitions(-DNABTO_RTSP_HAS_BASIC_AUTH)
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endif()
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if (RTSP_HAS_DIGEST_AUTH)
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add_definitions(-DNABTO_RTSP_HAS_DIGEST_AUTH)
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endif()
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add_executable(webrtc_device_rtsp_h265 "${src}")
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find_package(plog REQUIRED)
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find_package(cxxopts)
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find_package(LibDataChannel REQUIRED)
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find_package(CURL REQUIRED)
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find_package(NabtoWebrtcSignaling REQUIRED)
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find_package(OpenSSL REQUIRED)
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target_link_libraries(webrtc_device_rtsp_h265
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webrtc_example_common
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cxxopts::cxxopts
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LibDataChannel::LibDataChannel
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CURL::libcurl
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plog::plog
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NabtoWebrtcSignaling::device
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NabtoWebrtcSignaling::util_logging
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NabtoWebrtcSignaling::util_curl_client
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NabtoWebrtcSignaling::util_token_generator
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OpenSSL::Crypto
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)
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install(TARGETS webrtc_device_rtsp_h265
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RUNTIME DESTINATION bin
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)

‎examples/libdatachannel/src/webrtc_device_rtsp_h265/h265_opus_rtsp_handler.hpp‎

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#pragma once
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#include <memory>
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#include <nabto/webrtc/util/logging.hpp>
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#include <rtc/rtc.hpp>
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#include <webrtc_connection/track_handler.hpp>
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#include "rtsp-client/rtsp_client.hpp"
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namespace nabto {
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namespace example {
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class H265TrackHandler;
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typedef std::shared_ptr<H265TrackHandler> H265TrackHandlerPtr;
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class SsrcGenerator {
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public:
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static uint32_t generateSsrc() {
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static std::mutex mutex;
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std::lock_guard<std::mutex> lock(mutex);
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static uint32_t ssrc = 0;
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ssrc += 1;
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return ssrc;
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}
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};
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class MidGenerator {
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public:
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static std::string generateMid() {
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static std::mutex mutex;
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std::lock_guard<std::mutex> lock(mutex);
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static uint64_t midCounter = 0;
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std::stringstream ss;
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ss << "device-" << midCounter;
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midCounter += 1;
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return ss.str();
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}
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};
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class HandlerTrack {
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public:
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std::shared_ptr<rtc::Track> videoTrack;
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std::shared_ptr<rtc::Track> audioTrack;
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RtspClientPtr rtp;
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size_t ref;
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};
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class H265TrackHandler : public WebrtcTrackHandler,
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public std::enable_shared_from_this<H265TrackHandler> {
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public:
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static WebrtcTrackHandlerPtr create(std::string rtspUrl) {
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return std::make_shared<H265TrackHandler>(rtspUrl);
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}
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H265TrackHandler(std::string rtspUrl)
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: rtspUrl_(rtspUrl),
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ssrc_(SsrcGenerator::generateSsrc()),
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audioSsrc_(SsrcGenerator::generateSsrc()) {
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RtspClientConf conf = {rtspUrl, nullptr, nullptr, 96,
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111, ssrc_, audioSsrc_};
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rtp_ = RtspClient::create(conf);
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}
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size_t addTrack(std::shared_ptr<rtc::PeerConnection> pc) {
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counter_++;
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RtspClientConf conf = {
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rtspUrl_, nullptr, nullptr,
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96, 111, ssrc_,
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audioSsrc_, true, (uint16_t)(42222 + (counter_ * 4))};
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HandlerTrack track;
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track.rtp = RtspClient::create(conf);
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track.videoTrack = pc->addTrack(createVideoDescription());
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track.audioTrack = pc->addTrack(createAudioDescription());
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track.ref = track.rtp->addConnection(track.videoTrack, track.audioTrack);
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return track.ref;
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}
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void removeConnection(size_t ref) {
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auto conn = connections_.find(ref);
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if (conn == connections_.end()) {
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return;
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}
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conn->second.rtp->removeConnection(ref);
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connections_.erase(conn);
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}
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void close() {
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videoTrack_ = nullptr;
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rtp_ = nullptr;
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}
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private:
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std::shared_ptr<rtc::Track> videoTrack_;
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std::shared_ptr<rtc::Track> audioTrack_;
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RtspClientPtr rtp_;
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std::map<size_t, HandlerTrack> connections_;
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size_t counter_ = 0;
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std::string rtspUrl_;
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uint32_t ssrc_;
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uint32_t payloadType_ = 96;
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uint32_t audioSsrc_;
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rtc::Description::Video createVideoDescription() {
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// Create a Video media description.
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// We support both sending and receiving video
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std::string mid = MidGenerator::generateMid();
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rtc::Description::Video media(mid, rtc::Description::Direction::SendRecv);
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// media.addVideoCodec(
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// 96,
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// "m=video 0 RTP/AVP 96\r\nc=IN IP4 0.0.0.0\r\na=rtpmap:96 "
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// "H265/90000\r\na=framerate:30\r\na=fmtp:96 "
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// "sprop-vps=QAEMAf//"
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// "BAgAAAMAmAgAAAMAAFqSgJA=;sprop-sps="
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// "QgEBBAgAAAMAmAgAAAMAAFqQAKBAPCKUslSSZX/"
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// "4AAgAC1BgYGBAAAADAEAAAAeC;sprop-pps=RAHBcoYMRiQ=\r\na=control:"
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// "stream=0\r\na=ts-refclk:local\r\na=mediaclk:sender\r\na=ssrc:"
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// "1842951636 cname:user1268947453@host-65e41dfd\r\n");
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// Since we are creating the media track, only the supported payload
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// type exists, so we might as well reuse the same value for the RTP
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// session in WebRTC as the one we use in the RTP source (eg. Gstreamer)
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media.addH265Codec(payloadType_);
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auto r = media.rtpMap(96);
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// r->fmtps.push_back(
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// "sprop-vps=QAEMAf//"
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// "BAgAAAMAmAgAAAMAAFqSgJA=;sprop-sps="
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// "QgEBBAgAAAMAmAgAAAMAAFqQAKBAPCKUslSSZX/"
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// "4AAgAC1BgYGBAAAADAEAAAAeC;sprop-pps=RAHBcoYMRiQ=");
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return media;
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}
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rtc::Description::Audio createAudioDescription() {
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// Create an Audio media description.
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// We support both sending and receiving audio
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std::string mid = MidGenerator::generateMid();
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rtc::Description::Audio media(mid, rtc::Description::Direction::SendRecv);
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media.addOpusCodec(111);
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auto r = media.rtpMap(111);
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// media.addPCMUCodec(0);
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// auto r = media.rtpMap(0);
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r->removeFeedback("nack");
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r->removeFeedback("goog-remb");
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return media;
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}
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};
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} // namespace example
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} // namespace nabto

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