Add OpusRtpDepacketizer

Inverse of OpusRtpPacketizer. Takes incoming Opus packets and emits
frames.

CMakeLists.txt

@@ -75,6 +75,7 @@ set(LIBDATACHANNEL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtppacketizationconfig.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpsrreporter.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtppacketizer.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/rtpdepacketizer.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/h264rtppacketizer.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/h264rtpdepacketizer.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/nalunit.cpp
@@ -111,6 +112,7 @@ set(LIBDATACHANNEL_HEADERS
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtppacketizationconfig.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpsrreporter.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtppacketizer.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtpdepacketizer.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/h264rtppacketizer.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/h264rtpdepacketizer.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/nalunit.hpp

include/rtc/rtc.hpp

@@ -38,5 +38,6 @@
 #include "rtcpreceivingsession.hpp"
 #include "rtcpsrreporter.hpp"
 #include "rtppacketizer.hpp"
+#include "rtpdepacketizer.hpp"
 
 #endif // RTC_ENABLE_MEDIA

include/rtc/rtpdepacketizer.hpp

@@ -0,0 +1,34 @@
+/**
+ * Copyright (c) 2024 Paul-Louis Ageneau
+ *
+ * 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_RTP_DEPACKETIZER_H
+#define RTC_RTP_DEPACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "message.hpp"
+
+namespace rtc {
+
+class RTC_CPP_EXPORT RtpDepacketizer : public MediaHandler {
+public:
+	RtpDepacketizer() = default;
+	virtual ~RtpDepacketizer() = default;
+
+	virtual void incoming(message_vector &messages, const message_callback &send) override;
+};
+
+using OpusRtpDepacketizer = RtpDepacketizer;
+using AACRtpDepacketizer = RtpDepacketizer;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_DEPACKETIZER_H */

src/rtpdepacketizer.cpp

@@ -0,0 +1,47 @@
+/**
+ * Copyright (c) 2024 Paul-Louis Ageneau
+ *
+ * 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/.
+ */
+
+#if RTC_ENABLE_MEDIA
+
+#include "rtpdepacketizer.hpp"
+#include "rtp.hpp"
+
+#include "impl/logcounter.hpp"
+
+#include <cmath>
+#include <cstring>
+
+namespace rtc {
+
+void RtpDepacketizer::incoming([[maybe_unused]] message_vector &messages,
+                               [[maybe_unused]] const message_callback &send) {
+	message_vector result;
+	for (auto &message : messages) {
+		if (message->type == Message::Control) {
+			result.push_back(std::move(message));
+			continue;
+		}
+
+		if (message->size() < sizeof(RtpHeader)) {
+			PLOG_VERBOSE << "RTP packet is too small, size=" << message->size();
+			continue;
+		}
+
+		auto pkt = reinterpret_cast<const rtc::RtpHeader *>(message->data());
+		auto headerSize = sizeof(rtc::RtpHeader) + pkt->csrcCount() + pkt->getExtensionHeaderSize();
+		result.push_back(make_message(message->begin() + headerSize, message->end(),
+		                              Message::Binary, 0, nullptr,
+		                              std::make_shared<FrameInfo>(pkt->timestamp())));
+	}
+
+	messages.swap(result);
+}
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */