Haxe Lib Datachannel

lib/datachannel/cpp/libdatachannel/include/rtc/rtpdepacketizer.hpp

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+/**
+ * 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"
+
+#include <set>
+
+namespace rtc {
+
+// Base RTP depacketizer class
+class RTC_CPP_EXPORT RtpDepacketizer : public MediaHandler {
+public:
+	RtpDepacketizer();
+	RtpDepacketizer(uint32_t clockRate);
+	virtual ~RtpDepacketizer();
+
+	virtual void incoming(message_vector &messages, const message_callback &send) override;
+
+protected:
+	shared_ptr<FrameInfo> createFrameInfo(uint32_t timestamp, uint8_t payloadType) const;
+
+private:
+	const uint32_t mClockRate;
+};
+
+// Base class for video RTP depacketizer
+class RTC_CPP_EXPORT VideoRtpDepacketizer : public RtpDepacketizer {
+public:
+	inline static const uint32_t ClockRate = 90000;
+
+	VideoRtpDepacketizer();
+	virtual ~VideoRtpDepacketizer();
+
+protected:
+	struct sequence_cmp {
+		bool operator()(message_ptr a, message_ptr b) const;
+	};
+	using message_buffer = std::set<message_ptr, sequence_cmp>;
+
+	virtual message_ptr reassemble(message_buffer &messages) = 0;
+
+private:
+	void incoming(message_vector &messages, const message_callback &send) override;
+
+	message_buffer mBuffer;
+};
+
+// Generic audio RTP depacketizer
+template <uint32_t DEFAULT_CLOCK_RATE>
+class RTC_CPP_EXPORT AudioRtpDepacketizer final : public RtpDepacketizer {
+public:
+	inline static const uint32_t DefaultClockRate = DEFAULT_CLOCK_RATE;
+
+	AudioRtpDepacketizer(uint32_t clockRate = DefaultClockRate) : RtpDepacketizer(clockRate) {}
+};
+
+// Audio RTP depacketizers
+using OpusRtpDepacketizer = AudioRtpDepacketizer<48000>;
+using AACRtpDepacketizer = AudioRtpDepacketizer<48000>;
+using PCMARtpDepacketizer = AudioRtpDepacketizer<8000>;
+using PCMURtpDepacketizer = AudioRtpDepacketizer<8000>;
+using G722RtpDepacketizer = AudioRtpDepacketizer<8000>;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_DEPACKETIZER_H */