react-native-webrtc-nitro 1.0.0

package/3rdparty/output/android/libdatachannel/arm64-v8a/include/rtc/rtcpreceivingsession.hpp

@@ -0,0 +1,54 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 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_RTCP_RECEIVING_SESSION_H
+#define RTC_RTCP_RECEIVING_SESSION_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "common.hpp"
+#include "mediahandler.hpp"
+#include "message.hpp"
+#include "rtp.hpp"
+
+#include <atomic>
+
+namespace rtc {
+
+// An RtcpSession can be plugged into a Track to handle the whole RTCP session
+class RTC_CPP_EXPORT RtcpReceivingSession : public MediaHandler {
+public:
+	RtcpReceivingSession() = default;
+	virtual ~RtcpReceivingSession() = default;
+
+	void incoming(message_vector &messages, const message_callback &send) override;
+	bool requestKeyframe(const message_callback &send) override;
+	bool requestBitrate(unsigned int bitrate, const message_callback &send) override;
+
+	// For backward compatibility
+	[[deprecated("Use Track::requestKeyframe()")]] inline bool requestKeyframe() { return false; };
+	[[deprecated("Use Track::requestBitrate()")]] inline void requestBitrate(unsigned int) {};
+
+protected:
+	void pushREMB(const message_callback &send, unsigned int bitrate);
+	void pushRR(const message_callback &send,unsigned int lastSrDelay);
+	void pushPLI(const message_callback &send);
+
+	SSRC mSsrc = 0;
+	uint32_t mGreatestSeqNo = 0;
+	uint64_t mSyncRTPTS, mSyncNTPTS;
+
+	std::atomic<unsigned int> mRequestedBitrate = 0;
+};
+
+} // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_RTCP_RECEIVING_SESSION_H

package/3rdparty/output/android/libdatachannel/arm64-v8a/include/rtc/rtcpsrreporter.hpp

@@ -0,0 +1,49 @@
+/**
+ * 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_RTCP_SR_REPORTER_H
+#define RTC_RTCP_SR_REPORTER_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "mediahandler.hpp"
+#include "rtp.hpp"
+#include "rtppacketizationconfig.hpp"
+
+#include <chrono>
+
+namespace rtc {
+
+class RTC_CPP_EXPORT RtcpSrReporter final : public MediaHandler {
+public:
+	RtcpSrReporter(shared_ptr<RtpPacketizationConfig> rtpConfig);
+	~RtcpSrReporter();
+
+	uint32_t lastReportedTimestamp() const;
+	[[deprecated]] void setNeedsToReport();
+
+	void outgoing(message_vector &messages, const message_callback &send) override;
+
+	// TODO: remove this
+	const shared_ptr<RtpPacketizationConfig> rtpConfig;
+
+private:
+	void addToReport(RtpHeader *header, size_t size);
+	message_ptr getSenderReport(uint32_t timestamp);
+
+	uint32_t mPacketCount = 0;
+	uint32_t mPayloadOctets = 0;
+	uint32_t mLastReportedTimestamp = 0;
+	std::chrono::steady_clock::time_point mLastReportTime;
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTCP_SR_REPORTER_H */

package/3rdparty/output/android/libdatachannel/arm64-v8a/include/rtc/rtp.hpp

@@ -0,0 +1,367 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ * 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_HPP
+#define RTC_RTP_HPP
+
+#include "common.hpp"
+
+#include <vector>
+
+namespace rtc {
+
+typedef uint32_t SSRC;
+
+RTC_CPP_EXPORT bool IsRtcp(const binary &data);
+
+#pragma pack(push, 1)
+
+struct RTC_CPP_EXPORT RtpExtensionHeader {
+	uint16_t _profileSpecificId;
+	uint16_t _headerLength;
+
+	[[nodiscard]] uint16_t profileSpecificId() const;
+	[[nodiscard]] uint16_t headerLength() const;
+
+	[[nodiscard]] size_t getSize() const;
+	[[nodiscard]] const char *getBody() const;
+	[[nodiscard]] char *getBody();
+
+	void setProfileSpecificId(uint16_t profileSpecificId);
+	void setHeaderLength(uint16_t headerLength);
+
+	void clearBody();
+	size_t writeCurrentVideoOrientation(bool twoByteHeader, size_t offset, uint8_t id,
+	                                    uint8_t value);
+	size_t writeOneByteHeader(size_t offset, uint8_t id, const byte *value, size_t size);
+	size_t writeTwoByteHeader(size_t offset, uint8_t id, const byte *value, size_t size);
+	size_t writeHeader(bool twoByteHeader, size_t offset, uint8_t id, const byte *value,
+	                   size_t size);
+};
+
+struct RTC_CPP_EXPORT RtpHeader {
+	uint8_t _first;
+	uint8_t _payloadType;
+	uint16_t _seqNumber;
+	uint32_t _timestamp;
+	SSRC _ssrc;
+	// The following field is SSRC _csrc[]
+
+	[[nodiscard]] uint8_t version() const;
+	[[nodiscard]] bool padding() const;
+	[[nodiscard]] bool extension() const;
+	[[nodiscard]] uint8_t csrcCount() const;
+	[[nodiscard]] uint8_t marker() const;
+	[[nodiscard]] uint8_t payloadType() const;
+	[[nodiscard]] uint16_t seqNumber() const;
+	[[nodiscard]] uint32_t timestamp() const;
+	[[nodiscard]] uint32_t ssrc() const;
+
+	[[nodiscard]] size_t getSize() const;
+	[[nodiscard]] size_t getExtensionHeaderSize() const;
+	[[nodiscard]] const RtpExtensionHeader *getExtensionHeader() const;
+	[[nodiscard]] RtpExtensionHeader *getExtensionHeader();
+	[[nodiscard]] const char *getBody() const;
+	[[nodiscard]] char *getBody();
+
+	void log() const;
+
+	void preparePacket();
+	void setSeqNumber(uint16_t newSeqNo);
+	void setPayloadType(uint8_t newPayloadType);
+	void setSsrc(uint32_t in_ssrc);
+	void setMarker(bool marker);
+	void setTimestamp(uint32_t i);
+	void setExtension(bool extension);
+};
+
+struct RTC_CPP_EXPORT RtcpReportBlock {
+	SSRC _ssrc;
+	uint32_t _fractionLostAndPacketsLost; // fraction lost is 8-bit, packets lost is 24-bit
+	uint16_t _seqNoCycles;
+	uint16_t _highestSeqNo;
+	uint32_t _jitter;
+	uint32_t _lastReport;
+	uint32_t _delaySinceLastReport;
+
+	[[nodiscard]] uint16_t seqNoCycles() const;
+	[[nodiscard]] uint16_t highestSeqNo() const;
+	[[nodiscard]] uint32_t extendedHighestSeqNo() const;
+	[[nodiscard]] uint32_t jitter() const;
+	[[nodiscard]] uint32_t delaySinceSR() const;
+
+	[[nodiscard]] SSRC getSSRC() const;
+	[[nodiscard]] uint32_t getNTPOfSR() const;
+	[[nodiscard]] uint8_t getFractionLost() const;
+	[[nodiscard]] unsigned int getPacketsLostCount() const;
+
+	void preparePacket(SSRC in_ssrc, unsigned int packetsLost, unsigned int totalPackets,
+	                   uint16_t highestSeqNo, uint16_t seqNoCycles, uint32_t jitter,
+	                   uint64_t lastSR_NTP, uint64_t lastSR_DELAY);
+	void setSSRC(SSRC in_ssrc);
+	void setPacketsLost(uint8_t fractionLost, unsigned int packetsLostCount);
+	void setSeqNo(uint16_t highestSeqNo, uint16_t seqNoCycles);
+	void setJitter(uint32_t jitter);
+	void setNTPOfSR(uint64_t ntp);
+	void setDelaySinceSR(uint32_t sr);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpHeader {
+	uint8_t _first;
+	uint8_t _payloadType;
+	uint16_t _length;
+
+	[[nodiscard]] uint8_t version() const;
+	[[nodiscard]] bool padding() const;
+	[[nodiscard]] uint8_t reportCount() const;
+	[[nodiscard]] uint8_t payloadType() const;
+	[[nodiscard]] uint16_t length() const;
+	[[nodiscard]] size_t lengthInBytes() const;
+
+	void prepareHeader(uint8_t payloadType, uint8_t reportCount, uint16_t length);
+	void setPayloadType(uint8_t type);
+	void setReportCount(uint8_t count);
+	void setLength(uint16_t length);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpFbHeader {
+	RtcpHeader header;
+
+	SSRC _packetSender;
+	SSRC _mediaSource;
+
+	[[nodiscard]] SSRC packetSenderSSRC() const;
+	[[nodiscard]] SSRC mediaSourceSSRC() const;
+
+	void setPacketSenderSSRC(SSRC ssrc);
+	void setMediaSourceSSRC(SSRC ssrc);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpSr {
+	RtcpHeader header;
+
+	SSRC _senderSSRC;
+	uint64_t _ntpTimestamp;
+	uint32_t _rtpTimestamp;
+	uint32_t _packetCount;
+	uint32_t _octetCount;
+
+	RtcpReportBlock _reportBlocks;
+
+	[[nodiscard]] static unsigned int Size(unsigned int reportCount);
+
+	[[nodiscard]] uint64_t ntpTimestamp() const;
+	[[nodiscard]] uint32_t rtpTimestamp() const;
+	[[nodiscard]] uint32_t packetCount() const;
+	[[nodiscard]] uint32_t octetCount() const;
+	[[nodiscard]] uint32_t senderSSRC() const;
+
+	[[nodiscard]] const RtcpReportBlock *getReportBlock(int num) const;
+	[[nodiscard]] RtcpReportBlock *getReportBlock(int num);
+	[[nodiscard]] unsigned int size(unsigned int reportCount);
+	[[nodiscard]] size_t getSize() const;
+
+	void preparePacket(SSRC senderSSRC, uint8_t reportCount);
+	void setNtpTimestamp(uint64_t ts);
+	void setRtpTimestamp(uint32_t ts);
+	void setOctetCount(uint32_t ts);
+	void setPacketCount(uint32_t ts);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpSdesItem {
+	uint8_t type;
+
+	uint8_t _length;
+	char _text[1];
+
+	[[nodiscard]] static unsigned int Size(uint8_t textLength);
+
+	[[nodiscard]] string text() const;
+	[[nodiscard]] uint8_t length() const;
+
+	void setText(string text);
+};
+
+struct RTC_CPP_EXPORT RtcpSdesChunk {
+	SSRC _ssrc;
+	RtcpSdesItem _items;
+
+	[[nodiscard]] static unsigned int Size(const std::vector<uint8_t> textLengths);
+
+	[[nodiscard]] SSRC ssrc() const;
+
+	void setSSRC(SSRC ssrc);
+
+	// Get item at given index
+	// All items with index < num must be valid, otherwise this function has undefined behaviour
+	// (use safelyCountChunkSize() to check if chunk is valid).
+	[[nodiscard]] const RtcpSdesItem *getItem(int num) const;
+	[[nodiscard]] RtcpSdesItem *getItem(int num);
+
+	// Get size of chunk
+	// All items must be valid, otherwise this function has undefined behaviour (use
+	// safelyCountChunkSize() to check if chunk is valid)
+	[[nodiscard]] unsigned int getSize() const;
+
+	long safelyCountChunkSize(size_t maxChunkSize) const;
+};
+
+struct RTC_CPP_EXPORT RtcpSdes {
+	RtcpHeader header;
+	RtcpSdesChunk _chunks;
+
+	[[nodiscard]] static unsigned int Size(const std::vector<std::vector<uint8_t>> lengths);
+
+	bool isValid() const;
+
+	// Returns number of chunks in this packet
+	// Returns 0 if packet is invalid
+	unsigned int chunksCount() const;
+
+	// Get chunk at given index
+	// All chunks (and their items) with index < `num` must be valid, otherwise this function has
+	// undefined behaviour (use `isValid` to check if chunk is valid).
+	const RtcpSdesChunk *getChunk(int num) const;
+	RtcpSdesChunk *getChunk(int num);
+
+	void preparePacket(uint8_t chunkCount);
+};
+
+struct RTC_CPP_EXPORT RtcpRr {
+	RtcpHeader header;
+
+	SSRC _senderSSRC;
+	RtcpReportBlock _reportBlocks;
+
+	[[nodiscard]] static size_t SizeWithReportBlocks(uint8_t reportCount);
+
+	SSRC senderSSRC() const;
+	bool isSenderReport();
+	bool isReceiverReport();
+
+	[[nodiscard]] RtcpReportBlock *getReportBlock(int num);
+	[[nodiscard]] const RtcpReportBlock *getReportBlock(int num) const;
+	[[nodiscard]] size_t getSize() const;
+
+	void preparePacket(SSRC senderSSRC, uint8_t reportCount);
+	void setSenderSSRC(SSRC ssrc);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpRemb {
+	RtcpFbHeader header;
+
+	char _id[4];       // Unique identifier ('R' 'E' 'M' 'B')
+	uint32_t _bitrate; // Num SSRC, Br Exp, Br Mantissa (bit mask)
+	SSRC _ssrc[1];
+
+	[[nodiscard]] static size_t SizeWithSSRCs(int count);
+
+	[[nodiscard]] unsigned int getSize() const;
+
+	void preparePacket(SSRC senderSSRC, unsigned int numSSRC, unsigned int in_bitrate);
+	void setBitrate(unsigned int numSSRC, unsigned int in_bitrate);
+	void setSsrc(int iterator, SSRC newSssrc);
+	unsigned int getNumSSRC();
+	unsigned int getBitrate();
+};
+
+struct RTC_CPP_EXPORT RtcpPli {
+	RtcpFbHeader header;
+
+	[[nodiscard]] static unsigned int Size();
+
+	void preparePacket(SSRC messageSSRC);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpFirPart {
+	uint32_t ssrc;
+	uint8_t seqNo;
+	uint8_t dummy1;
+	uint16_t dummy2;
+};
+
+struct RTC_CPP_EXPORT RtcpFir {
+	RtcpFbHeader header;
+	RtcpFirPart parts[1];
+
+	static unsigned int Size();
+
+	void preparePacket(SSRC messageSSRC, uint8_t seqNo);
+
+	void log() const;
+};
+
+struct RTC_CPP_EXPORT RtcpNackPart {
+	uint16_t _pid;
+	uint16_t _blp;
+
+	uint16_t pid();
+	uint16_t blp();
+
+	void setPid(uint16_t pid);
+	void setBlp(uint16_t blp);
+
+	std::vector<uint16_t> getSequenceNumbers();
+};
+
+struct RTC_CPP_EXPORT RtcpNack {
+	RtcpFbHeader header;
+	RtcpNackPart parts[1];
+
+	[[nodiscard]] static unsigned int Size(unsigned int discreteSeqNoCount);
+
+	[[nodiscard]] unsigned int getSeqNoCount();
+
+	void preparePacket(SSRC ssrc, unsigned int discreteSeqNoCount);
+
+	/**
+	 * Add a packet to the list of missing packets.
+	 * @param fciCount The number of FCI fields that are present in this packet.
+	 *                  Let the number start at zero and let this function grow the number.
+	 * @param fciPID The seq no of the active FCI. It will be initialized automatically, and will
+	 * change automatically.
+	 * @param missingPacket The seq no of the missing packet. This will be added to the queue.
+	 * @return true if the packet has grown, false otherwise.
+	 */
+	bool addMissingPacket(unsigned int *fciCount, uint16_t *fciPID, uint16_t missingPacket);
+};
+
+struct RTC_CPP_EXPORT RtpRtx {
+	RtpHeader header;
+
+	[[nodiscard]] const char *getBody() const;
+	[[nodiscard]] char *getBody();
+	[[nodiscard]] size_t getBodySize(size_t totalSize) const;
+	[[nodiscard]] size_t getSize() const;
+	[[nodiscard]] uint16_t getOriginalSeqNo() const;
+
+	// Returns the new size of the packet
+	size_t normalizePacket(size_t totalSize, SSRC originalSSRC, uint8_t originalPayloadType);
+
+	size_t copyTo(RtpHeader *dest, size_t totalSize, uint8_t originalPayloadType);
+};
+
+#pragma pack(pop)
+
+} // namespace rtc
+
+#endif

package/3rdparty/output/android/libdatachannel/arm64-v8a/include/rtc/rtpdepacketizer.hpp

@@ -0,0 +1,79 @@
+/**
+ * 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 */

package/3rdparty/output/android/libdatachannel/arm64-v8a/include/rtc/rtppacketizationconfig.hpp

@@ -0,0 +1,112 @@
+/**
+ * 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_PACKETIZATION_CONFIG_H
+#define RTC_RTP_PACKETIZATION_CONFIG_H
+
+#if RTC_ENABLE_MEDIA
+
+#include "dependencydescriptor.hpp"
+#include "rtp.hpp"
+
+namespace rtc {
+
+// RTP configuration used in packetization process
+class RTC_CPP_EXPORT RtpPacketizationConfig {
+public:
+	SSRC ssrc;
+	std::string cname;
+	uint8_t payloadType;
+	uint32_t clockRate;
+	uint8_t videoOrientationId;
+
+	// current sequence number
+	uint16_t sequenceNumber;
+
+	// current timestamp
+	uint32_t timestamp;
+
+	// start timestamp
+	uint32_t startTimestamp;
+
+	/// Current video orientation
+	///
+	/// Bit#       7  6  5  4  3  2  1  0
+	/// Definition 0  0  0  0  C  F  R1 R0
+	///
+	/// C
+	///   0 - Front-facing camera (use this if unsure)
+	///   1 - Back-facing camera
+	///
+	/// F
+	///   0 - No Flip
+	///   1 - Horizontal flip
+	///
+	/// R1 R0 - CW rotation that receiver must apply
+	///   0 - 0 degrees
+	///   1 - 90 degrees
+	///   2 - 180 degrees
+	///   3 - 270 degrees
+	uint8_t videoOrientation = 0;
+
+	// MID Extension Header
+	uint8_t midId = 0;
+	optional<std::string> mid;
+
+	// RID Extension Header
+	uint8_t ridId = 0;
+	optional<std::string> rid;
+
+	// Dependency Descriptor Extension Header
+	uint8_t dependencyDescriptorId = 0;
+
+	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 = 0;
+
+	// Minimum/maxiumum playout delay, in 10ms intervals. A value of 10 would equal a 100ms delay
+	uint16_t playoutDelayMin = 0;
+	uint16_t playoutDelayMax = 0;
+
+	/// Construct RTP configuration used in packetization process
+	/// @param ssrc SSRC of source
+	/// @param cname CNAME of source
+	/// @param payloadType Payload type of source
+	/// @param clockRate Clock rate of source used in timestamps
+	/// nullopt)
+	/// @param videoOrientationId Video orientation (see above)
+	RtpPacketizationConfig(SSRC ssrc, std::string cname, uint8_t payloadType, uint32_t clockRate,
+	                       uint8_t videoOrientationId = 0);
+
+	RtpPacketizationConfig(const RtpPacketizationConfig &) = delete;
+
+	/// Convert timestamp to seconds
+	/// @param timestamp Timestamp
+	/// @param clockRate Clock rate for timestamp calculation
+	static double getSecondsFromTimestamp(uint32_t timestamp, uint32_t clockRate);
+
+	/// Convert timestamp to seconds
+	/// @param timestamp Timestamp
+	double timestampToSeconds(uint32_t timestamp);
+
+	/// Convert seconds to timestamp
+	/// @param seconds Number of seconds
+	/// @param clockRate Clock rate for timestamp calculation
+	static uint32_t getTimestampFromSeconds(double seconds, uint32_t clockRate);
+
+	/// Convert seconds to timestamp
+	/// @param seconds Number of seconds
+	uint32_t secondsToTimestamp(double seconds);
+};
+
+} // namespace rtc
+
+#endif /* RTC_ENABLE_MEDIA */
+
+#endif /* RTC_RTP_PACKETIZATION_CONFIG_H */