diff options
Diffstat (limited to 'webrtc/modules/rtp_rtcp/include')
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/fec_receiver.h | 46 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/receive_statistics.h | 102 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h | 51 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/rtp_cvo.h | 54 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/rtp_header_parser.h | 44 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h | 203 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/rtp_receiver.h | 103 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/rtp_rtcp.h | 653 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h | 416 |
9 files changed, 1672 insertions, 0 deletions
diff --git a/webrtc/modules/rtp_rtcp/include/fec_receiver.h b/webrtc/modules/rtp_rtcp/include/fec_receiver.h new file mode 100644 index 0000000000..65e85ad7a5 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/fec_receiver.h @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct FecPacketCounter { + FecPacketCounter() + : num_packets(0), + num_fec_packets(0), + num_recovered_packets(0) {} + + size_t num_packets; // Number of received packets. + size_t num_fec_packets; // Number of received FEC packets. + size_t num_recovered_packets; // Number of recovered media packets using FEC. +}; + +class FecReceiver { + public: + static FecReceiver* Create(RtpData* callback); + + virtual ~FecReceiver() {} + + virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header, + const uint8_t* incoming_rtp_packet, + size_t packet_length, + uint8_t ulpfec_payload_type) = 0; + + virtual int32_t ProcessReceivedFec() = 0; + + virtual FecPacketCounter GetPacketCounter() const = 0; +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ diff --git a/webrtc/modules/rtp_rtcp/include/receive_statistics.h b/webrtc/modules/rtp_rtcp/include/receive_statistics.h new file mode 100644 index 0000000000..b4a7cd0de2 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/receive_statistics.h @@ -0,0 +1,102 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ + +#include <map> + +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class Clock; + +class StreamStatistician { + public: + virtual ~StreamStatistician(); + + virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0; + virtual void GetDataCounters(size_t* bytes_received, + uint32_t* packets_received) const = 0; + + // Gets received stream data counters (includes reset counter values). + virtual void GetReceiveStreamDataCounters( + StreamDataCounters* data_counters) const = 0; + + virtual uint32_t BitrateReceived() const = 0; + + // Returns true if the packet with RTP header |header| is likely to be a + // retransmitted packet, false otherwise. + virtual bool IsRetransmitOfOldPacket(const RTPHeader& header, + int64_t min_rtt) const = 0; + + // Returns true if |sequence_number| is received in order, false otherwise. + virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0; +}; + +typedef std::map<uint32_t, StreamStatistician*> StatisticianMap; + +class ReceiveStatistics : public Module { + public: + virtual ~ReceiveStatistics() {} + + static ReceiveStatistics* Create(Clock* clock); + + // Updates the receive statistics with this packet. + virtual void IncomingPacket(const RTPHeader& rtp_header, + size_t packet_length, + bool retransmitted) = 0; + + // Increment counter for number of FEC packets received. + virtual void FecPacketReceived(const RTPHeader& header, + size_t packet_length) = 0; + + // Returns a map of all statisticians which have seen an incoming packet + // during the last two seconds. + virtual StatisticianMap GetActiveStatisticians() const = 0; + + // Returns a pointer to the statistician of an ssrc. + virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0; + + // Sets the max reordering threshold in number of packets. + virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0; + + // Called on new RTCP stats creation. + virtual void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) = 0; + + // Called on new RTP stats creation. + virtual void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) = 0; +}; + +class NullReceiveStatistics : public ReceiveStatistics { + public: + void IncomingPacket(const RTPHeader& rtp_header, + size_t packet_length, + bool retransmitted) override; + void FecPacketReceived(const RTPHeader& header, + size_t packet_length) override; + StatisticianMap GetActiveStatisticians() const override; + StreamStatistician* GetStatistician(uint32_t ssrc) const override; + int64_t TimeUntilNextProcess() override; + int32_t Process() override; + void SetMaxReorderingThreshold(int max_reordering_threshold) override; + void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) override; + void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) override; +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ diff --git a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h new file mode 100644 index 0000000000..56c6e48691 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/system_wrappers/include/rtp_to_ntp.h" + +namespace webrtc { + +class Clock; +class TimestampExtrapolator; + +// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP +// time in local timebase. +// Note that it needs to be trained with at least 2 RTCP SR (by calling +// |UpdateRtcpTimestamp|) before it can be used. +class RemoteNtpTimeEstimator { + public: + explicit RemoteNtpTimeEstimator(Clock* clock); + + ~RemoteNtpTimeEstimator(); + + // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|, + // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|. + bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac, + uint32_t rtp_timestamp); + + // Estimates the NTP timestamp in local timebase from |rtp_timestamp|. + // Returns the NTP timestamp in ms when success. -1 if failed. + int64_t Estimate(uint32_t rtp_timestamp); + + private: + Clock* clock_; + rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_; + RtcpList rtcp_list_; + int64_t last_timing_log_ms_; + RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_cvo.h b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h new file mode 100644 index 0000000000..2e30d898ec --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h @@ -0,0 +1,54 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ + +#include "webrtc/common_video/rotation.h" + +namespace webrtc { + +// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/ +// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is +// the clockwise angle the frames must be rotated in order to display the frames +// correctly if the display is rotated in its natural orientation. +inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) { + switch (rotation) { + case kVideoRotation_0: + return 0; + case kVideoRotation_90: + return 1; + case kVideoRotation_180: + return 2; + case kVideoRotation_270: + return 3; + } + assert(false); + return 0; +} + +inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) { + switch (rotation) { + case 0: + return kVideoRotation_0; + case 1: + return kVideoRotation_90; + case 2: + return kVideoRotation_180; + break; + case 3: + return kVideoRotation_270; + default: + assert(false); + return kVideoRotation_0; + } +} + +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h new file mode 100644 index 0000000000..329de32611 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h @@ -0,0 +1,44 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct RTPHeader; + +class RtpHeaderParser { + public: + static RtpHeaderParser* Create(); + virtual ~RtpHeaderParser() {} + + // Returns true if the packet is an RTCP packet, false otherwise. + static bool IsRtcp(const uint8_t* packet, size_t length); + + // Parses the packet and stores the parsed packet in |header|. Returns true on + // success, false otherwise. + // This method is thread-safe in the sense that it can parse multiple packets + // at once. + virtual bool Parse(const uint8_t* packet, + size_t length, + RTPHeader* header) const = 0; + + // Registers an RTP header extension and binds it to |id|. + virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, + uint8_t id) = 0; + + // De-registers an RTP header extension. + virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0; +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h new file mode 100644 index 0000000000..fae864107f --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h @@ -0,0 +1,203 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ + +#include <map> + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" + +namespace webrtc { + +// This strategy deals with the audio/video-specific aspects +// of payload handling. +class RTPPayloadStrategy { + public: + virtual ~RTPPayloadStrategy() {} + + virtual bool CodecsMustBeUnique() const = 0; + + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const size_t channels, + const uint32_t rate) const = 0; + + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const = 0; + + virtual RtpUtility::Payload* CreatePayloadType( + const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t frequency, + const size_t channels, + const uint32_t rate) const = 0; + + virtual int GetPayloadTypeFrequency( + const RtpUtility::Payload& payload) const = 0; + + static RTPPayloadStrategy* CreateStrategy(const bool handling_audio); + + protected: + RTPPayloadStrategy() {} +}; + +class RTPPayloadRegistry { + public: + // The registry takes ownership of the strategy. + explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy); + ~RTPPayloadRegistry(); + + int32_t RegisterReceivePayload( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const int8_t payload_type, + const uint32_t frequency, + const size_t channels, + const uint32_t rate, + bool* created_new_payload_type); + + int32_t DeRegisterReceivePayload( + const int8_t payload_type); + + int32_t ReceivePayloadType( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const uint32_t frequency, + const size_t channels, + const uint32_t rate, + int8_t* payload_type) const; + + bool RtxEnabled() const; + + void SetRtxSsrc(uint32_t ssrc); + + bool GetRtxSsrc(uint32_t* ssrc) const; + + void SetRtxPayloadType(int payload_type, int associated_payload_type); + + bool IsRtx(const RTPHeader& header) const; + + // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t* + // restored_packet, instead of a uint8_t**. + // TODO(noahric): Remove this when all callers have been updated. + bool RestoreOriginalPacket(uint8_t** restored_packet, + const uint8_t* packet, + size_t* packet_length, + uint32_t original_ssrc, + const RTPHeader& header) const; + + bool RestoreOriginalPacket(uint8_t* restored_packet, + const uint8_t* packet, + size_t* packet_length, + uint32_t original_ssrc, + const RTPHeader& header) const; + + bool IsRed(const RTPHeader& header) const; + + // Returns true if the media of this RTP packet is encapsulated within an + // extra header, such as RTX or RED. + bool IsEncapsulated(const RTPHeader& header) const; + + bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const; + + int GetPayloadTypeFrequency(uint8_t payload_type) const; + + // DEPRECATED. Use PayloadTypeToPayload below that returns const Payload* + // instead of taking output parameter. + // TODO(danilchap): Remove this when all callers have been updated. + bool PayloadTypeToPayload(const uint8_t payload_type, + RtpUtility::Payload*& payload) const { // NOLINT + payload = + const_cast<RtpUtility::Payload*>(PayloadTypeToPayload(payload_type)); + return payload != nullptr; + } + const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const; + + void ResetLastReceivedPayloadTypes() { + CriticalSectionScoped cs(crit_sect_.get()); + last_received_payload_type_ = -1; + last_received_media_payload_type_ = -1; + } + + // This sets the payload type of the packets being received from the network + // on the media SSRC. For instance if packets are encapsulated with RED, this + // payload type will be the RED payload type. + void SetIncomingPayloadType(const RTPHeader& header); + + // Returns true if the new media payload type has not changed. + bool ReportMediaPayloadType(uint8_t media_payload_type); + + int8_t red_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return red_payload_type_; + } + int8_t ulpfec_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return ulpfec_payload_type_; + } + int8_t last_received_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return last_received_payload_type_; + } + void set_last_received_payload_type(int8_t last_received_payload_type) { + CriticalSectionScoped cs(crit_sect_.get()); + last_received_payload_type_ = last_received_payload_type; + } + + int8_t last_received_media_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return last_received_media_payload_type_; + } + + bool use_rtx_payload_mapping_on_restore() const { + CriticalSectionScoped cs(crit_sect_.get()); + return use_rtx_payload_mapping_on_restore_; + } + + void set_use_rtx_payload_mapping_on_restore(bool val) { + CriticalSectionScoped cs(crit_sect_.get()); + use_rtx_payload_mapping_on_restore_ = val; + } + + private: + // Prunes the payload type map of the specific payload type, if it exists. + void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const size_t payload_name_length, + const uint32_t frequency, + const size_t channels, + const uint32_t rate); + + bool IsRtxInternal(const RTPHeader& header) const; + + rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_; + RtpUtility::PayloadTypeMap payload_type_map_; + rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_; + int8_t red_payload_type_; + int8_t ulpfec_payload_type_; + int8_t incoming_payload_type_; + int8_t last_received_payload_type_; + int8_t last_received_media_payload_type_; + bool rtx_; + // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that + // only understand one RTX PT is no longer needed. + int rtx_payload_type_; + // Mapping rtx_payload_type_map_[rtx] = associated. + std::map<int, int> rtx_payload_type_map_; + // When true, use rtx_payload_type_map_ when restoring RTX packets to get the + // correct payload type. + bool use_rtx_payload_mapping_on_restore_; + uint32_t ssrc_rtx_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h new file mode 100644 index 0000000000..0640d5cc19 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h @@ -0,0 +1,103 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class RTPPayloadRegistry; + +class TelephoneEventHandler { + public: + virtual ~TelephoneEventHandler() {} + + // The following three methods implement the TelephoneEventHandler interface. + // Forward DTMFs to decoder for playout. + virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0; + + // Is forwarding of outband telephone events turned on/off? + virtual bool TelephoneEventForwardToDecoder() const = 0; + + // Is TelephoneEvent configured with payload type payload_type + virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0; +}; + +class RtpReceiver { + public: + // Creates a video-enabled RTP receiver. + static RtpReceiver* CreateVideoReceiver( + Clock* clock, + RtpData* incoming_payload_callback, + RtpFeedback* incoming_messages_callback, + RTPPayloadRegistry* rtp_payload_registry); + + // Creates an audio-enabled RTP receiver. + static RtpReceiver* CreateAudioReceiver( + Clock* clock, + RtpAudioFeedback* incoming_audio_feedback, + RtpData* incoming_payload_callback, + RtpFeedback* incoming_messages_callback, + RTPPayloadRegistry* rtp_payload_registry); + + virtual ~RtpReceiver() {} + + // Returns a TelephoneEventHandler if available. + virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; + + // Registers a receive payload in the payload registry and notifies the media + // receiver strategy. + virtual int32_t RegisterReceivePayload( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const int8_t payload_type, + const uint32_t frequency, + const size_t channels, + const uint32_t rate) = 0; + + // De-registers |payload_type| from the payload registry. + virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0; + + // Parses the media specific parts of an RTP packet and updates the receiver + // state. This for instance means that any changes in SSRC and payload type is + // detected and acted upon. + virtual bool IncomingRtpPacket(const RTPHeader& rtp_header, + const uint8_t* payload, + size_t payload_length, + PayloadUnion payload_specific, + bool in_order) = 0; + + // Returns the currently configured NACK method. + virtual NACKMethod NACK() const = 0; + + // Turn negative acknowledgement (NACK) requests on/off. + virtual void SetNACKStatus(const NACKMethod method) = 0; + + // Gets the last received timestamp. Returns true if a packet has been + // received, false otherwise. + virtual bool Timestamp(uint32_t* timestamp) const = 0; + // Gets the time in milliseconds when the last timestamp was received. + // Returns true if a packet has been received, false otherwise. + virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0; + + // Returns the remote SSRC of the currently received RTP stream. + virtual uint32_t SSRC() const = 0; + + // Returns the current remote CSRCs. + virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0; + + // Returns the current energy of the RTP stream received. + virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0; +}; +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h new file mode 100644 index 0000000000..6a7022a94c --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h @@ -0,0 +1,653 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ + +#include <set> +#include <utility> +#include <vector> + +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" + +namespace webrtc { +// Forward declarations. +class ReceiveStatistics; +class RemoteBitrateEstimator; +class RtpReceiver; +class Transport; +namespace rtcp { +class TransportFeedback; +} + +class RtpRtcp : public Module { + public: + struct Configuration { + Configuration(); + + /* id - Unique identifier of this RTP/RTCP module object + * audio - True for a audio version of the RTP/RTCP module + * object false will create a video version + * clock - The clock to use to read time. If NULL object + * will be using the system clock. + * incoming_data - Callback object that will receive the incoming + * data. May not be NULL; default callback will do + * nothing. + * incoming_messages - Callback object that will receive the incoming + * RTP messages. May not be NULL; default callback + * will do nothing. + * outgoing_transport - Transport object that will be called when packets + * are ready to be sent out on the network + * intra_frame_callback - Called when the receiver request a intra frame. + * bandwidth_callback - Called when we receive a changed estimate from + * the receiver of out stream. + * audio_messages - Telephone events. May not be NULL; default + * callback will do nothing. + * remote_bitrate_estimator - Estimates the bandwidth available for a set of + * streams from the same client. + * paced_sender - Spread any bursts of packets into smaller + * bursts to minimize packet loss. + */ + bool audio; + bool receiver_only; + Clock* clock; + ReceiveStatistics* receive_statistics; + Transport* outgoing_transport; + RtcpIntraFrameObserver* intra_frame_callback; + RtcpBandwidthObserver* bandwidth_callback; + TransportFeedbackObserver* transport_feedback_callback; + RtcpRttStats* rtt_stats; + RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; + RtpAudioFeedback* audio_messages; + RemoteBitrateEstimator* remote_bitrate_estimator; + RtpPacketSender* paced_sender; + TransportSequenceNumberAllocator* transport_sequence_number_allocator; + BitrateStatisticsObserver* send_bitrate_observer; + FrameCountObserver* send_frame_count_observer; + SendSideDelayObserver* send_side_delay_observer; + }; + + /* + * Create a RTP/RTCP module object using the system clock. + * + * configuration - Configuration of the RTP/RTCP module. + */ + static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); + + /************************************************************************** + * + * Receiver functions + * + ***************************************************************************/ + + virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, + size_t incoming_packet_length) = 0; + + virtual void SetRemoteSSRC(uint32_t ssrc) = 0; + + /************************************************************************** + * + * Sender + * + ***************************************************************************/ + + /* + * set MTU + * + * size - Max transfer unit in bytes, default is 1500 + * + * return -1 on failure else 0 + */ + virtual int32_t SetMaxTransferUnit(uint16_t size) = 0; + + /* + * set transtport overhead + * default is IPv4 and UDP with no encryption + * + * TCP - true for TCP false UDP + * IPv6 - true for IP version 6 false for version 4 + * authenticationOverhead - number of bytes to leave for an + * authentication header + * + * return -1 on failure else 0 + */ + virtual int32_t SetTransportOverhead( + bool TCP, + bool IPV6, + uint8_t authenticationOverhead = 0) = 0; + + /* + * Get max payload length + * + * A combination of the configuration MaxTransferUnit and + * TransportOverhead. + * Does not account FEC/ULP/RED overhead if FEC is enabled. + * Does not account for RTP headers + */ + virtual uint16_t MaxPayloadLength() const = 0; + + /* + * Get max data payload length + * + * A combination of the configuration MaxTransferUnit, headers and + * TransportOverhead. + * Takes into account FEC/ULP/RED overhead if FEC is enabled. + * Takes into account RTP headers + */ + virtual uint16_t MaxDataPayloadLength() const = 0; + + /* + * set codec name and payload type + * + * return -1 on failure else 0 + */ + virtual int32_t RegisterSendPayload( + const CodecInst& voiceCodec) = 0; + + /* + * set codec name and payload type + * + * return -1 on failure else 0 + */ + virtual int32_t RegisterSendPayload( + const VideoCodec& videoCodec) = 0; + + /* + * Unregister a send payload + * + * payloadType - payload type of codec + * + * return -1 on failure else 0 + */ + virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0; + + /* + * (De)register RTP header extension type and id. + * + * return -1 on failure else 0 + */ + virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type, + uint8_t id) = 0; + + virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; + + /* + * get start timestamp + */ + virtual uint32_t StartTimestamp() const = 0; + + /* + * configure start timestamp, default is a random number + * + * timestamp - start timestamp + */ + virtual void SetStartTimestamp(uint32_t timestamp) = 0; + + /* + * Get SequenceNumber + */ + virtual uint16_t SequenceNumber() const = 0; + + /* + * Set SequenceNumber, default is a random number + */ + virtual void SetSequenceNumber(uint16_t seq) = 0; + + // Returns true if the ssrc matched this module, false otherwise. + virtual bool SetRtpStateForSsrc(uint32_t ssrc, + const RtpState& rtp_state) = 0; + virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0; + + /* + * Get SSRC + */ + virtual uint32_t SSRC() const = 0; + + /* + * configure SSRC, default is a random number + */ + virtual void SetSSRC(uint32_t ssrc) = 0; + + /* + * Set CSRC + * + * csrcs - vector of CSRCs + */ + virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0; + + /* + * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination + * of values of the enumerator RtxMode. + */ + virtual void SetRtxSendStatus(int modes) = 0; + + /* + * Get status of sending RTX (RFC 4588). The returned value can be + * a combination of values of the enumerator RtxMode. + */ + virtual int RtxSendStatus() const = 0; + + // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, + // only the SSRC is set. + virtual void SetRtxSsrc(uint32_t ssrc) = 0; + + // Sets the payload type to use when sending RTX packets. Note that this + // doesn't enable RTX, only the payload type is set. + virtual void SetRtxSendPayloadType(int payload_type, + int associated_payload_type) = 0; + + // Gets the payload type pair of (RTX, associated) to use when sending RTX + // packets. + virtual std::pair<int, int> RtxSendPayloadType() const = 0; + + /* + * sends kRtcpByeCode when going from true to false + * + * sending - on/off + * + * return -1 on failure else 0 + */ + virtual int32_t SetSendingStatus(bool sending) = 0; + + /* + * get send status + */ + virtual bool Sending() const = 0; + + /* + * Starts/Stops media packets, on by default + * + * sending - on/off + */ + virtual void SetSendingMediaStatus(bool sending) = 0; + + /* + * get send status + */ + virtual bool SendingMedia() const = 0; + + /* + * get sent bitrate in Kbit/s + */ + virtual void BitrateSent(uint32_t* totalRate, + uint32_t* videoRate, + uint32_t* fecRate, + uint32_t* nackRate) const = 0; + + /* + * Used by the codec module to deliver a video or audio frame for + * packetization. + * + * frameType - type of frame to send + * payloadType - payload type of frame to send + * timestamp - timestamp of frame to send + * payloadData - payload buffer of frame to send + * payloadSize - size of payload buffer to send + * fragmentation - fragmentation offset data for fragmented frames such + * as layers or RED + * + * return -1 on failure else 0 + */ + virtual int32_t SendOutgoingData( + FrameType frameType, + int8_t payloadType, + uint32_t timeStamp, + int64_t capture_time_ms, + const uint8_t* payloadData, + size_t payloadSize, + const RTPFragmentationHeader* fragmentation = NULL, + const RTPVideoHeader* rtpVideoHdr = NULL) = 0; + + virtual bool TimeToSendPacket(uint32_t ssrc, + uint16_t sequence_number, + int64_t capture_time_ms, + bool retransmission) = 0; + + virtual size_t TimeToSendPadding(size_t bytes) = 0; + + // Called on generation of new statistics after an RTP send. + virtual void RegisterSendChannelRtpStatisticsCallback( + StreamDataCountersCallback* callback) = 0; + virtual StreamDataCountersCallback* + GetSendChannelRtpStatisticsCallback() const = 0; + + /************************************************************************** + * + * RTCP + * + ***************************************************************************/ + + /* + * Get RTCP status + */ + virtual RtcpMode RTCP() const = 0; + + /* + * configure RTCP status i.e on(compound or non- compound)/off + * + * method - RTCP method to use + */ + virtual void SetRTCPStatus(RtcpMode method) = 0; + + /* + * Set RTCP CName (i.e unique identifier) + * + * return -1 on failure else 0 + */ + virtual int32_t SetCNAME(const char* c_name) = 0; + + /* + * Get remote CName + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteCNAME(uint32_t remoteSSRC, + char cName[RTCP_CNAME_SIZE]) const = 0; + + /* + * Get remote NTP + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteNTP( + uint32_t *ReceivedNTPsecs, + uint32_t *ReceivedNTPfrac, + uint32_t *RTCPArrivalTimeSecs, + uint32_t *RTCPArrivalTimeFrac, + uint32_t *rtcp_timestamp) const = 0; + + /* + * AddMixedCNAME + * + * return -1 on failure else 0 + */ + virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0; + + /* + * RemoveMixedCNAME + * + * return -1 on failure else 0 + */ + virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0; + + /* + * Get RoundTripTime + * + * return -1 on failure else 0 + */ + virtual int32_t RTT(uint32_t remoteSSRC, + int64_t* RTT, + int64_t* avgRTT, + int64_t* minRTT, + int64_t* maxRTT) const = 0; + + /* + * Force a send of a RTCP packet + * periodic SR and RR are triggered via the process function + * + * return -1 on failure else 0 + */ + virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0; + + /* + * Force a send of a RTCP packet with more than one packet type. + * periodic SR and RR are triggered via the process function + * + * return -1 on failure else 0 + */ + virtual int32_t SendCompoundRTCP( + const std::set<RTCPPacketType>& rtcpPacketTypes) = 0; + + /* + * Good state of RTP receiver inform sender + */ + virtual int32_t SendRTCPReferencePictureSelection( + const uint64_t pictureID) = 0; + + /* + * Send a RTCP Slice Loss Indication (SLI) + * 6 least significant bits of pictureID + */ + virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0; + + /* + * Statistics of the amount of data sent + * + * return -1 on failure else 0 + */ + virtual int32_t DataCountersRTP( + size_t* bytesSent, + uint32_t* packetsSent) const = 0; + + /* + * Get send statistics for the RTP and RTX stream. + */ + virtual void GetSendStreamDataCounters( + StreamDataCounters* rtp_counters, + StreamDataCounters* rtx_counters) const = 0; + + /* + * Get packet loss statistics for the RTP stream. + */ + virtual void GetRtpPacketLossStats( + bool outgoing, + uint32_t ssrc, + struct RtpPacketLossStats* loss_stats) const = 0; + + /* + * Get received RTCP sender info + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0; + + /* + * Get received RTCP report block + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteRTCPStat( + std::vector<RTCPReportBlock>* receiveBlocks) const = 0; + + /* + * (APP) Application specific data + * + * return -1 on failure else 0 + */ + virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType, + uint32_t name, + const uint8_t* data, + uint16_t length) = 0; + /* + * (XR) VOIP metric + * + * return -1 on failure else 0 + */ + virtual int32_t SetRTCPVoIPMetrics( + const RTCPVoIPMetric* VoIPMetric) = 0; + + /* + * (XR) Receiver Reference Time Report + */ + virtual void SetRtcpXrRrtrStatus(bool enable) = 0; + + virtual bool RtcpXrRrtrStatus() const = 0; + + /* + * (REMB) Receiver Estimated Max Bitrate + */ + virtual bool REMB() const = 0; + + virtual void SetREMBStatus(bool enable) = 0; + + virtual void SetREMBData(uint32_t bitrate, + const std::vector<uint32_t>& ssrcs) = 0; + + /* + * (TMMBR) Temporary Max Media Bit Rate + */ + virtual bool TMMBR() const = 0; + + virtual void SetTMMBRStatus(bool enable) = 0; + + /* + * (NACK) + */ + + /* + * TODO(holmer): Propagate this API to VideoEngine. + * Returns the currently configured selective retransmission settings. + */ + virtual int SelectiveRetransmissions() const = 0; + + /* + * TODO(holmer): Propagate this API to VideoEngine. + * Sets the selective retransmission settings, which will decide which + * packets will be retransmitted if NACKed. Settings are constructed by + * combining the constants in enum RetransmissionMode with bitwise OR. + * All packets are retransmitted if kRetransmitAllPackets is set, while no + * packets are retransmitted if kRetransmitOff is set. + * By default all packets except FEC packets are retransmitted. For VP8 + * with temporal scalability only base layer packets are retransmitted. + * + * Returns -1 on failure, otherwise 0. + */ + virtual int SetSelectiveRetransmissions(uint8_t settings) = 0; + + /* + * Send a Negative acknowledgement packet + * + * return -1 on failure else 0 + */ + virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0; + + /* + * Store the sent packets, needed to answer to a Negative acknowledgement + * requests + */ + virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0; + + // Returns true if the module is configured to store packets. + virtual bool StorePackets() const = 0; + + // Called on receipt of RTCP report block from remote side. + virtual void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) = 0; + virtual RtcpStatisticsCallback* + GetRtcpStatisticsCallback() = 0; + // BWE feedback packets. + virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; + + /************************************************************************** + * + * Audio + * + ***************************************************************************/ + + /* + * set audio packet size, used to determine when it's time to send a DTMF + * packet in silence (CNG) + * + * return -1 on failure else 0 + */ + virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0; + + /* + * Send a TelephoneEvent tone using RFC 2833 (4733) + * + * return -1 on failure else 0 + */ + virtual int32_t SendTelephoneEventOutband(uint8_t key, + uint16_t time_ms, + uint8_t level) = 0; + + /* + * Set payload type for Redundant Audio Data RFC 2198 + * + * return -1 on failure else 0 + */ + virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0; + + /* + * Get payload type for Redundant Audio Data RFC 2198 + * + * return -1 on failure else 0 + */ + // DEPRECATED. Use SendREDPayloadType below that takes output parameter + // by pointer instead of by reference. + // TODO(danilchap): Remove this when all callers have been updated. + int32_t SendREDPayloadType(int8_t& payloadType) const { // NOLINT + return SendREDPayloadType(&payloadType); + } + virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 0; + /* + * Store the audio level in dBov for header-extension-for-audio-level- + * indication. + * This API shall be called before transmision of an RTP packet to ensure + * that the |level| part of the extended RTP header is updated. + * + * return -1 on failure else 0. + */ + virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0; + + /************************************************************************** + * + * Video + * + ***************************************************************************/ + + /* + * Set the target send bitrate + */ + virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0; + + /* + * Turn on/off generic FEC + */ + virtual void SetGenericFECStatus(bool enable, + uint8_t payload_type_red, + uint8_t payload_type_fec) = 0; + + /* + * Get generic FEC setting + */ + // DEPRECATED. Use GenericFECStatus below that takes output parameters + // by pointers instead of by references. + // TODO(danilchap): Remove this when all callers have been updated. + void GenericFECStatus(bool& enable, // NOLINT + uint8_t& payloadTypeRED, // NOLINT + uint8_t& payloadTypeFEC) { // NOLINT + GenericFECStatus(&enable, &payloadTypeRED, &payloadTypeFEC); + } + virtual void GenericFECStatus(bool* enable, + uint8_t* payload_type_red, + uint8_t* payload_type_fec) = 0; + + virtual int32_t SetFecParameters( + const FecProtectionParams* delta_params, + const FecProtectionParams* key_params) = 0; + + /* + * Set method for requestion a new key frame + * + * return -1 on failure else 0 + */ + virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; + + /* + * send a request for a keyframe + * + * return -1 on failure else 0 + */ + virtual int32_t RequestKeyFrame() = 0; +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h new file mode 100644 index 0000000000..fad97f19cc --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h @@ -0,0 +1,416 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ + +#include <stddef.h> +#include <list> + +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/system_wrappers/include/clock.h" +#include "webrtc/typedefs.h" + +#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination +#define IP_PACKET_SIZE 1500 // we assume ethernet +#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 +#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds + +namespace webrtc { +namespace rtcp { +class TransportFeedback; +} + +const int kVideoPayloadTypeFrequency = 90000; + +// Minimum RTP header size in bytes. +const uint8_t kRtpHeaderSize = 12; + +struct AudioPayload { + uint32_t frequency; + size_t channels; + uint32_t rate; +}; + +struct VideoPayload { + RtpVideoCodecTypes videoCodecType; + uint32_t maxRate; +}; + +union PayloadUnion { + AudioPayload Audio; + VideoPayload Video; +}; + +enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; + +enum ProtectionType { + kUnprotectedPacket, + kProtectedPacket +}; + +enum StorageType { + kDontRetransmit, + kAllowRetransmission +}; + +enum RTPExtensionType { + kRtpExtensionNone, + kRtpExtensionTransmissionTimeOffset, + kRtpExtensionAudioLevel, + kRtpExtensionAbsoluteSendTime, + kRtpExtensionVideoRotation, + kRtpExtensionTransportSequenceNumber, +}; + +enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 }; + +// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up. +enum RTCPPacketType : uint32_t { + kRtcpReport = 0x0001, + kRtcpSr = 0x0002, + kRtcpRr = 0x0004, + kRtcpSdes = 0x0008, + kRtcpBye = 0x0010, + kRtcpPli = 0x0020, + kRtcpNack = 0x0040, + kRtcpFir = 0x0080, + kRtcpTmmbr = 0x0100, + kRtcpTmmbn = 0x0200, + kRtcpSrReq = 0x0400, + kRtcpXrVoipMetric = 0x0800, + kRtcpApp = 0x1000, + kRtcpSli = 0x4000, + kRtcpRpsi = 0x8000, + kRtcpRemb = 0x10000, + kRtcpTransmissionTimeOffset = 0x20000, + kRtcpXrReceiverReferenceTime = 0x40000, + kRtcpXrDlrrReportBlock = 0x80000, + kRtcpTransportFeedback = 0x100000, +}; + +enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp }; + +enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 }; + +enum NACKMethod { kNackOff = 0, kNackRtcp = 2 }; + +enum RetransmissionMode : uint8_t { + kRetransmitOff = 0x0, + kRetransmitFECPackets = 0x1, + kRetransmitBaseLayer = 0x2, + kRetransmitHigherLayers = 0x4, + kRetransmitAllPackets = 0xFF +}; + +enum RtxMode { + kRtxOff = 0x0, + kRtxRetransmitted = 0x1, // Only send retransmissions over RTX. + kRtxRedundantPayloads = 0x2 // Preventively send redundant payloads + // instead of padding. +}; + +const size_t kRtxHeaderSize = 2; + +struct RTCPSenderInfo { + uint32_t NTPseconds; + uint32_t NTPfraction; + uint32_t RTPtimeStamp; + uint32_t sendPacketCount; + uint32_t sendOctetCount; +}; + +struct RTCPReportBlock { + RTCPReportBlock() + : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0), + extendedHighSeqNum(0), jitter(0), lastSR(0), + delaySinceLastSR(0) {} + + RTCPReportBlock(uint32_t remote_ssrc, + uint32_t source_ssrc, + uint8_t fraction_lost, + uint32_t cumulative_lost, + uint32_t extended_high_sequence_number, + uint32_t jitter, + uint32_t last_sender_report, + uint32_t delay_since_last_sender_report) + : remoteSSRC(remote_ssrc), + sourceSSRC(source_ssrc), + fractionLost(fraction_lost), + cumulativeLost(cumulative_lost), + extendedHighSeqNum(extended_high_sequence_number), + jitter(jitter), + lastSR(last_sender_report), + delaySinceLastSR(delay_since_last_sender_report) {} + + // Fields as described by RFC 3550 6.4.2. + uint32_t remoteSSRC; // SSRC of sender of this report. + uint32_t sourceSSRC; // SSRC of the RTP packet sender. + uint8_t fractionLost; + uint32_t cumulativeLost; // 24 bits valid. + uint32_t extendedHighSeqNum; + uint32_t jitter; + uint32_t lastSR; + uint32_t delaySinceLastSR; +}; + +struct RtcpReceiveTimeInfo { + // Fields as described by RFC 3611 4.5. + uint32_t sourceSSRC; + uint32_t lastRR; + uint32_t delaySinceLastRR; +}; + +typedef std::list<RTCPReportBlock> ReportBlockList; + +struct RtpState { + RtpState() + : sequence_number(0), + start_timestamp(0), + timestamp(0), + capture_time_ms(-1), + last_timestamp_time_ms(-1), + media_has_been_sent(false) {} + uint16_t sequence_number; + uint32_t start_timestamp; + uint32_t timestamp; + int64_t capture_time_ms; + int64_t last_timestamp_time_ms; + bool media_has_been_sent; +}; + +class RtpData { + public: + virtual ~RtpData() {} + + virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData, + const size_t payloadSize, + const WebRtcRTPHeader* rtpHeader) = 0; + + virtual bool OnRecoveredPacket(const uint8_t* packet, + size_t packet_length) = 0; +}; + +class RtpFeedback { + public: + virtual ~RtpFeedback() {} + + // Receiving payload change or SSRC change. (return success!) + /* + * channels - number of channels in codec (1 = mono, 2 = stereo) + */ + virtual int32_t OnInitializeDecoder( + const int8_t payloadType, + const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int frequency, + const size_t channels, + const uint32_t rate) = 0; + + virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0; + + virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0; +}; + +class RtpAudioFeedback { + public: + virtual void OnPlayTelephoneEvent(const uint8_t event, + const uint16_t lengthMs, + const uint8_t volume) = 0; + + protected: + virtual ~RtpAudioFeedback() {} +}; + +class RtcpIntraFrameObserver { + public: + virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0; + + virtual void OnReceivedSLI(uint32_t ssrc, + uint8_t picture_id) = 0; + + virtual void OnReceivedRPSI(uint32_t ssrc, + uint64_t picture_id) = 0; + + virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0; + + virtual ~RtcpIntraFrameObserver() {} +}; + +class RtcpBandwidthObserver { + public: + // REMB or TMMBR + virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0; + + virtual void OnReceivedRtcpReceiverReport( + const ReportBlockList& report_blocks, + int64_t rtt, + int64_t now_ms) = 0; + + virtual ~RtcpBandwidthObserver() {} +}; + +struct PacketInfo { + PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number) + : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {} + + PacketInfo(int64_t arrival_time_ms, + int64_t send_time_ms, + uint16_t sequence_number, + size_t payload_size, + bool was_paced) + : PacketInfo(-1, + arrival_time_ms, + send_time_ms, + sequence_number, + payload_size, + was_paced) {} + + PacketInfo(int64_t creation_time_ms, + int64_t arrival_time_ms, + int64_t send_time_ms, + uint16_t sequence_number, + size_t payload_size, + bool was_paced) + : creation_time_ms(creation_time_ms), + arrival_time_ms(arrival_time_ms), + send_time_ms(send_time_ms), + sequence_number(sequence_number), + payload_size(payload_size), + was_paced(was_paced) {} + + // Time corresponding to when this object was created. + int64_t creation_time_ms; + // Time corresponding to when the packet was received. Timestamped with the + // receiver's clock. + int64_t arrival_time_ms; + // Time corresponding to when the packet was sent, timestamped with the + // sender's clock. + int64_t send_time_ms; + // Packet identifier, incremented with 1 for every packet generated by the + // sender. + uint16_t sequence_number; + // Size of the packet excluding RTP headers. + size_t payload_size; + // True if the packet was paced out by the pacer. + bool was_paced; +}; + +class TransportFeedbackObserver { + public: + TransportFeedbackObserver() {} + virtual ~TransportFeedbackObserver() {} + + // Note: Transport-wide sequence number as sequence number. Arrival time + // must be set to 0. + virtual void AddPacket(uint16_t sequence_number, + size_t length, + bool was_paced) = 0; + + virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0; +}; + +class RtcpRttStats { + public: + virtual void OnRttUpdate(int64_t rtt) = 0; + + virtual int64_t LastProcessedRtt() const = 0; + + virtual ~RtcpRttStats() {} +}; + +// Null object version of RtpFeedback. +class NullRtpFeedback : public RtpFeedback { + public: + virtual ~NullRtpFeedback() {} + + int32_t OnInitializeDecoder(const int8_t payloadType, + const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int frequency, + const size_t channels, + const uint32_t rate) override { + return 0; + } + + void OnIncomingSSRCChanged(const uint32_t ssrc) override {} + void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {} +}; + +// Null object version of RtpData. +class NullRtpData : public RtpData { + public: + virtual ~NullRtpData() {} + + int32_t OnReceivedPayloadData(const uint8_t* payloadData, + const size_t payloadSize, + const WebRtcRTPHeader* rtpHeader) override { + return 0; + } + + bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override { + return true; + } +}; + +// Null object version of RtpAudioFeedback. +class NullRtpAudioFeedback : public RtpAudioFeedback { + public: + virtual ~NullRtpAudioFeedback() {} + + void OnPlayTelephoneEvent(const uint8_t event, + const uint16_t lengthMs, + const uint8_t volume) override {} +}; + +// Statistics about packet loss for a single directional connection. All values +// are totals since the connection initiated. +struct RtpPacketLossStats { + // The number of packets lost in events where no adjacent packets were also + // lost. + uint64_t single_packet_loss_count; + // The number of events in which more than one adjacent packet was lost. + uint64_t multiple_packet_loss_event_count; + // The number of packets lost in events where more than one adjacent packet + // was lost. + uint64_t multiple_packet_loss_packet_count; +}; + +class RtpPacketSender { + public: + RtpPacketSender() {} + virtual ~RtpPacketSender() {} + + enum Priority { + kHighPriority = 0, // Pass through; will be sent immediately. + kNormalPriority = 2, // Put in back of the line. + kLowPriority = 3, // Put in back of the low priority line. + }; + // Low priority packets are mixed with the normal priority packets + // while we are paused. + + // Returns true if we send the packet now, else it will add the packet + // information to the queue and call TimeToSendPacket when it's time to send. + virtual void InsertPacket(Priority priority, + uint32_t ssrc, + uint16_t sequence_number, + int64_t capture_time_ms, + size_t bytes, + bool retransmission) = 0; +}; + +class TransportSequenceNumberAllocator { + public: + TransportSequenceNumberAllocator() {} + virtual ~TransportSequenceNumberAllocator() {} + + virtual uint16_t AllocateSequenceNumber() = 0; +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ |