Merge remote-tracking branch 'origin/master' into dependency-descriptor

Author: melpon

CMakeLists.txt

@@ -1,6 +1,6 @@
 cmake_minimum_required(VERSION 3.7)
 project(libdatachannel
-	VERSION 0.20.2
+	VERSION 0.21.0
 	LANGUAGES CXX)
 set(PROJECT_DESCRIPTION "C/C++ WebRTC network library featuring Data Channels, Media Transport, and WebSockets")
 
@@ -88,6 +88,7 @@ set(LIBDATACHANNEL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtp.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/capi.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/plihandler.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/pacinghandler.cpp
 )
 
 set(LIBDATACHANNEL_HEADERS
@@ -125,6 +126,7 @@ set(LIBDATACHANNEL_HEADERS
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpnackresponder.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/utils.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/plihandler.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/pacinghandler.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/version.h
 )
 
@@ -344,7 +346,9 @@ else()
 		target_link_libraries(datachannel PRIVATE libSRTP::srtp2)
 		target_link_libraries(datachannel-static PRIVATE libSRTP::srtp2)
 	else()
-		add_subdirectory(deps/libsrtp EXCLUDE_FROM_ALL)
+		if(NOT TARGET srtp2)
+			add_subdirectory(deps/libsrtp EXCLUDE_FROM_ALL)
+		endif()
 		target_compile_definitions(datachannel PRIVATE RTC_SYSTEM_SRTP=0)
 		target_compile_definitions(datachannel-static PRIVATE RTC_SYSTEM_SRTP=0)
 		target_link_libraries(datachannel PRIVATE srtp2)
@@ -373,7 +377,9 @@ if (USE_GNUTLS)
 		target_link_libraries(datachannel-static PRIVATE Nettle::Nettle)
 	endif()
 elseif(USE_MBEDTLS)
-	find_package(MbedTLS 3 REQUIRED)
+	if(NOT TARGET MbedTLS::MbedTLS)
+		find_package(MbedTLS 3 REQUIRED)
+	endif()
 	target_compile_definitions(datachannel PRIVATE USE_MBEDTLS=1)
 	target_compile_definitions(datachannel-static PRIVATE USE_MBEDTLS=1)
 	target_link_libraries(datachannel PRIVATE MbedTLS::MbedTLS)

deps/libjuice

@@ -11,6 +11,7 @@
 #include "rtc/rtc.hpp"
 
 #include <fstream>
+#include <cstring>
 
 #ifdef _WIN32
 #include <winsock2.h>
@@ -29,7 +30,9 @@ void H264FileParser::loadNextSample() {
     while (i < sample.size()) {
         assert(i + 4 < sample.size());
         auto lengthPtr = (uint32_t *) (sample.data() + i);
-        uint32_t length = ntohl(*lengthPtr);
+        uint32_t length;
+        std::memcpy(&length, lengthPtr, sizeof(uint32_t));
+        length = ntohl(length);
         auto naluStartIndex = i + 4;
         auto naluEndIndex = naluStartIndex + length;
         assert(naluEndIndex <= sample.size());
@@ -50,8 +53,8 @@ void H264FileParser::loadNextSample() {
     }
 }
 
-vector<byte> H264FileParser::initialNALUS() {
-    vector<byte> units{};
+vector<std::byte> H264FileParser::initialNALUS() {
+    vector<std::byte> units{};
     if (previousUnitType7.has_value()) {
         auto nalu = previousUnitType7.value();
         units.insert(units.end(), nalu.begin(), nalu.end());

examples/streamer/h264fileparser.cpp

@@ -75,6 +75,7 @@ struct RTC_CPP_EXPORT Configuration {
 	bool enableIceTcp = false;    // libnice only
 	bool enableIceUdpMux = false; // libjuice only
 	bool disableAutoNegotiation = false;
+	bool disableAutoGathering = false;
 	bool forceMediaTransport = false;
 
 	// Port range

include/rtc/configuration.hpp

@@ -14,7 +14,8 @@
 namespace rtc {
 
 struct RTC_CPP_EXPORT FrameInfo {
-	FrameInfo(uint32_t timestamp) : timestamp(timestamp){};
+	FrameInfo(uint8_t payloadType, uint32_t timestamp) : payloadType(payloadType), timestamp(timestamp){};
+	uint8_t payloadType; // Indicates codec of the frame
 	uint32_t timestamp = 0; // RTP Timestamp
 };
 

include/rtc/frameinfo.hpp

@@ -15,6 +15,7 @@
 #include "common.hpp"
 #include "mediahandler.hpp"
 #include "message.hpp"
+#include "nalunit.hpp"
 #include "rtp.hpp"
 
 #include <iterator>
@@ -24,16 +25,20 @@ namespace rtc {
 /// RTP depacketization for H264
 class RTC_CPP_EXPORT H264RtpDepacketizer : public MediaHandler {
 public:
-	H264RtpDepacketizer() = default;
+	using Separator = NalUnit::Separator;
+
+	H264RtpDepacketizer(Separator separator = Separator::LongStartSequence);
 	virtual ~H264RtpDepacketizer() = default;
 
 	void incoming(message_vector &messages, const message_callback &send) override;
 
 private:
 	std::vector<message_ptr> mRtpBuffer;
+	const NalUnit::Separator mSeparator;
 
+	void addSeparator(binary &accessUnit);
 	message_vector buildFrames(message_vector::iterator firstPkt, message_vector::iterator lastPkt,
-	                           uint32_t timestamp);
+	                           uint8_t payloadType, uint32_t timestamp);
 };
 
 } // namespace rtc

include/rtc/h264rtpdepacketizer.hpp

@@ -0,0 +1,49 @@
+/**
+ * Copyright (c) 2024 Sean DuBois <sean@siobud.com>
+ *
+ * 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/.
+ */
+
+#ifndef RTC_PACING_HANDLER_H
+#define RTC_PACING_HANDLER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "utils.hpp"
+
+#include <atomic>
+#include <queue>
+
+namespace rtc {
+
+// Paced sending of RTP packets. It takes a stream of RTP packets that can have an uneven bitrate
+// and delivers them in a smoother manner by sending a fixed size of them on an interval
+class RTC_CPP_EXPORT PacingHandler : public MediaHandler {
+public:
+	PacingHandler(double bitsPerSecond, std::chrono::milliseconds sendInterval);
+
+	void outgoing(message_vector &messages, const message_callback &send) override;
+
+private:
+	std::atomic<bool> mHaveScheduled = false;
+
+	double mBytesPerSecond;
+	double mBudget;
+
+	std::chrono::milliseconds mSendInterval;
+	std::chrono::time_point<std::chrono::high_resolution_clock> mLastRun;
+
+	std::mutex mMutex;
+	std::queue<message_ptr> mRtpBuffer;
+
+	void schedule(const message_callback &send);
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_PACING_HANDLER_H

include/rtc/pacinghandler.hpp

@@ -93,6 +93,7 @@ class RTC_CPP_EXPORT PeerConnection final : CheshireCat<impl::PeerConnection> {
 	void setLocalDescription(Description::Type type = Description::Type::Unspec);
 	void setRemoteDescription(Description description);
 	void addRemoteCandidate(Candidate candidate);
+	void gatherLocalCandidates(std::vector<IceServer> additionalIceServers = {});
 
 	void setMediaHandler(shared_ptr<MediaHandler> handler);
 	shared_ptr<MediaHandler> getMediaHandler();

include/rtc/peerconnection.hpp

@@ -341,6 +341,9 @@ typedef struct {
 	// AV1 only
 	rtcObuPacketization obuPacketization; // OBU paketization for AV1 samples
 
+	uint8_t playoutDelayId;
+	uint16_t playoutDelayMin;
+	uint16_t playoutDelayMax;
 } rtcPacketizerInit;
 
 // Deprecated, do not use

include/rtc/rtc.h

@@ -35,6 +35,7 @@
 #include "h265rtppacketizer.hpp"
 #include "mediahandler.hpp"
 #include "plihandler.hpp"
+#include "pacinghandler.hpp"
 #include "rtcpnackresponder.hpp"
 #include "rtcpreceivingsession.hpp"
 #include "rtcpsrreporter.hpp"

include/rtc/rtc.hpp

@@ -358,25 +358,6 @@ struct RTC_CPP_EXPORT RtpRtx {
 	size_t copyTo(RtpHeader *dest, size_t totalSize, uint8_t originalPayloadType);
 };
 
-// For backward compatibility, do not use
-using RTP_ExtensionHeader [[deprecated]] = RtpExtensionHeader;
-using RTP [[deprecated]] = RtpHeader;
-using RTCP_ReportBlock [[deprecated]] = RtcpReportBlock;
-using RTCP_HEADER [[deprecated]] = RtcpHeader;
-using RTCP_FB_HEADER [[deprecated]] = RtcpFbHeader;
-using RTCP_SR [[deprecated]] = RtcpSr;
-using RTCP_SDES_ITEM [[deprecated]] = RtcpSdesItem;
-using RTCP_SDES_CHUNK [[deprecated]] = RtcpSdesChunk;
-using RTCP_SDES [[deprecated]] = RtcpSdes;
-using RTCP_RR [[deprecated]] = RtcpRr;
-using RTCP_REMB [[deprecated]] = RtcpRemb;
-using RTCP_PLI [[deprecated]] = RtcpPli;
-using RTCP_FIR_PART [[deprecated]] = RtcpFirPart;
-using RTCP_FIR [[deprecated]] = RtcpFir;
-using RTCP_NACK_PART [[deprecated]] = RtcpNackPart;
-using RTCP_NACK [[deprecated]] = RtcpNack;
-using RTP_RTX [[deprecated]] = RtpRtx;
-
 #pragma pack(pop)
 
 } // namespace rtc

include/rtc/rtp.hpp

@@ -69,7 +69,15 @@ class RTC_CPP_EXPORT RtpPacketizationConfig {
 		std::bitset<32> activeChains;
 		FrameDependencyStructure structure;
 	};
+
 	optional<DependencyDescriptorContext> dependencyDescriptorContext;
+	// the negotiated ID of the playout delay header extension
+	// https://webrtc.googlesource.com/src/+/main/docs/native-code/rtp-hdrext/playout-delay/README.md
+	uint8_t playoutDelayId;
+
+	// Minimum/maxiumum playout delay, in 10ms intervals. A value of 10 would equal a 100ms delay
+	uint16_t playoutDelayMin;
+	uint16_t playoutDelayMax;
 
 	/// Construct RTP configuration used in packetization process
 	/// @param ssrc SSRC of source

include/rtc/rtppacketizationconfig.hpp

@@ -2,8 +2,8 @@
 #define RTC_VERSION_H
 
 #define RTC_VERSION_MAJOR 0
-#define RTC_VERSION_MINOR 20
-#define RTC_VERSION_PATCH 2
-#define RTC_VERSION "0.20.2"
+#define RTC_VERSION_MINOR 21
+#define RTC_VERSION_PATCH 0
+#define RTC_VERSION "0.21.0"
 
 #endif

include/rtc/version.h

@@ -275,6 +275,9 @@ createRtpPacketizationConfig(const rtcPacketizationHandlerInit *init) {
 	                                                       init->payloadType, init->clockRate);
 	config->sequenceNumber = init->sequenceNumber;
 	config->timestamp = init->timestamp;
+	config->playoutDelayId = init->playoutDelayId;
+	config->playoutDelayMin = init->playoutDelayMin;
+	config->playoutDelayMax = init->playoutDelayMax;
 	return config;
 }