Add OpusRtpDepacketizer

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

Author: Sean-Der

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
@@ -110,6 +111,7 @@ 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 +1,32 @@
+/**
+ * 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 {
+
+/// RTP depacketization for Opus
+class RTC_CPP_EXPORT OpusRtpDepacketizer : public MediaHandler {
+public:
+	OpusRtpDepacketizer() = default;
+	virtual ~OpusRtpDepacketizer() = default;
+
+	virtual void incoming(message_vector &messages, const message_callback &send) override;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_DEPACKETIZER_H */

src/rtpdepacketizer.cpp

@@ -0,0 +1,50 @@
+/**
+ * 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 OpusRtpDepacketizer::incoming([[maybe_unused]] message_vector &messages,
+                                   [[maybe_unused]] const message_callback &send) {
+
+	messages.erase(std::remove_if(messages.begin(), messages.end(),
+	                              [](message_ptr message) {
+		                              // Filter RTCP
+		                              if (message->type == Message::Control) {
+			                              return true;
+		                              }
+
+		                              if (message->size() < sizeof(RtpHeader)) {
+			                              PLOG_VERBOSE << "RTP packet is too small, size="
+			                                           << message->size();
+			                              return true;
+		                              }
+
+		                              return false;
+	                              }),
+	               messages.end());
+
+	for (auto &message : messages) {
+		auto pkt = reinterpret_cast<const rtc::RtpHeader *>(message->data());
+		auto headerSize = sizeof(rtc::RtpHeader) + pkt->csrcCount() + pkt->getExtensionHeaderSize();
+		message->erase(message->begin(), message->begin() + headerSize);
+	}
+}
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */