Haxe Lib Datachannel

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

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+/**
+ * Copyright (c) 2020 Filip Klembara (in2core)
+ *
+ * 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_PACKETIZER_H
+#define RTC_RTP_PACKETIZER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "message.hpp"
+#include "rtppacketizationconfig.hpp"
+
+namespace rtc {
+
+/// RTP packetizer
+class RTC_CPP_EXPORT RtpPacketizer : public MediaHandler {
+public:
+	/// Default maximum fragment size (for video packetizers)
+	inline static const size_t DefaultMaxFragmentSize = RTC_DEFAULT_MAX_FRAGMENT_SIZE;
+
+	/// Clock rate for video in RTP
+	inline static const uint32_t VideoClockRate = 90 * 1000;
+
+	/// Constructs packetizer with given RTP configuration
+	/// @note RTP configuration is used in packetization process which may change some configuration
+	/// properties such as sequence number.
+	/// @param rtpConfig  RTP configuration
+	RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig);
+	virtual ~RtpPacketizer();
+
+	virtual void media(const Description::Media &desc) override;
+	virtual void outgoing(message_vector &messages, const message_callback &send) override;
+
+	/// RTP packetization config
+	const shared_ptr<RtpPacketizationConfig> rtpConfig;
+
+protected:
+	/// Fragment data into payloads
+	/// Default implementation returns data as a single payload
+	/// @param data Input data
+	virtual std::vector<binary> fragment(binary data);
+
+	/// Creates an RTP packet for a payload
+	/// @note This function increases the sequence number.
+	/// @param payload RTP payload
+	/// @param mark Set marker flag in RTP packet if true
+	virtual message_ptr packetize(const binary &payload, bool mark, shared_ptr<FrameInfo> frameInfo = nullptr);
+
+private:
+	static const auto RtpHeaderSize = 12;
+	static const auto RtpExtHeaderCvoSize = 8;
+
+	uint32_t videoLayersAllocationInitialPacketCount = 0;
+
+	bool shouldEmitVideoLayersAllocation(shared_ptr<FrameInfo> frameInfo);
+};
+
+// Generic audio RTP packetizer
+template <uint32_t DEFAULT_CLOCK_RATE>
+class RTC_CPP_EXPORT AudioRtpPacketizer final : public RtpPacketizer {
+public:
+	inline static const uint32_t DefaultClockRate = DEFAULT_CLOCK_RATE;
+	inline static const uint32_t defaultClockRate [[deprecated("Use DefaultClockRate")]] =
+	    DEFAULT_CLOCK_RATE; // for backward compatibility
+
+	AudioRtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig)
+	    : RtpPacketizer(std::move(rtpConfig)) {}
+};
+
+// Audio RTP packetizers
+using OpusRtpPacketizer = AudioRtpPacketizer<48000>;
+using AACRtpPacketizer = AudioRtpPacketizer<48000>;
+using PCMARtpPacketizer = AudioRtpPacketizer<8000>;
+using PCMURtpPacketizer = AudioRtpPacketizer<8000>;
+using G722RtpPacketizer = AudioRtpPacketizer<8000>;
+
+// Dummy wrapper for backward compatibility, do not use
+class RTC_CPP_EXPORT PacketizationHandler final : public MediaHandler {
+public:
+	PacketizationHandler(shared_ptr<RtpPacketizer> packetizer)
+	    : mPacketizer(std::move(packetizer)) {}
+
+	inline void outgoing(message_vector &messages, const message_callback &send) {
+		return mPacketizer->outgoing(messages, send);
+	}
+
+private:
+	shared_ptr<RtpPacketizer> mPacketizer;
+};
+
+// Audio packetization handlers for backward compatibility, do not use
+using OpusPacketizationHandler [[deprecated("Add OpusRtpPacketizer directly")]] =
+    PacketizationHandler;
+using AACPacketizationHandler [[deprecated("Add AACRtpPacketizer directly")]] =
+    PacketizationHandler;
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_PACKETIZER_H */