| /* |
| * 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_SOURCE_RTP_RECEIVER_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_ |
| |
| #include <map> |
| |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/bitrate.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class RtpRtcpFeedback; |
| class ModuleRtpRtcpImpl; |
| class Trace; |
| class RTPReceiverAudio; |
| class RTPReceiverVideo; |
| class RTPReceiverStrategy; |
| |
| class RTPReceiver : public Bitrate { |
| public: |
| // Callbacks passed in here may not be NULL (use Null Object callbacks if you |
| // want callbacks to do nothing). This class takes ownership of the media |
| // receiver but nothing else. |
| RTPReceiver(const int32_t id, |
| Clock* clock, |
| ModuleRtpRtcpImpl* owner, |
| RtpAudioFeedback* incoming_audio_messages_callback, |
| RtpData* incoming_payload_callback, |
| RtpFeedback* incoming_messages_callback, |
| RTPReceiverStrategy* rtp_media_receiver, |
| RTPPayloadRegistry* rtp_payload_registry); |
| |
| virtual ~RTPReceiver(); |
| |
| RtpVideoCodecTypes VideoCodecType() const; |
| uint32_t MaxConfiguredBitrate() const; |
| |
| int32_t SetPacketTimeout(const uint32_t timeout_ms); |
| void PacketTimeout(); |
| |
| void ProcessDeadOrAlive(const bool RTCPalive, const int64_t now); |
| |
| void ProcessBitrate(); |
| |
| int32_t RegisterReceivePayload( |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const int8_t payload_type, |
| const uint32_t frequency, |
| const uint8_t channels, |
| const uint32_t rate); |
| |
| int32_t DeRegisterReceivePayload(const int8_t payload_type); |
| |
| int32_t ReceivePayloadType( |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const uint32_t frequency, |
| const uint8_t channels, |
| const uint32_t rate, |
| int8_t* payload_type) const; |
| |
| int32_t IncomingRTPPacket( |
| WebRtcRTPHeader* rtpheader, |
| const uint8_t* incoming_rtp_packet, |
| const uint16_t incoming_rtp_packet_length); |
| |
| NACKMethod NACK() const ; |
| |
| // Turn negative acknowledgement requests on/off. |
| int32_t SetNACKStatus(const NACKMethod method, int max_reordering_threshold); |
| |
| // Returns the last received timestamp. |
| virtual uint32_t TimeStamp() const; |
| int32_t LastReceivedTimeMs() const; |
| virtual uint16_t SequenceNumber() const; |
| |
| int32_t EstimatedRemoteTimeStamp(uint32_t& timestamp) const; |
| |
| uint32_t SSRC() const; |
| |
| int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const; |
| |
| int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const; |
| |
| // Get the currently configured SSRC filter. |
| int32_t SSRCFilter(uint32_t& allowed_ssrc) const; |
| |
| // Set a SSRC to be used as a filter for incoming RTP streams. |
| int32_t SetSSRCFilter(const bool enable, const uint32_t allowed_ssrc); |
| |
| int32_t Statistics(uint8_t* fraction_lost, |
| uint32_t* cum_lost, |
| uint32_t* ext_max, |
| uint32_t* jitter, // Will be moved from JB. |
| uint32_t* max_jitter, |
| uint32_t* jitter_transmission_time_offset, |
| bool reset) const; |
| |
| int32_t Statistics(uint8_t* fraction_lost, |
| uint32_t* cum_lost, |
| uint32_t* ext_max, |
| uint32_t* jitter, // Will be moved from JB. |
| uint32_t* max_jitter, |
| uint32_t* jitter_transmission_time_offset, |
| int32_t* missing, |
| bool reset) const; |
| |
| int32_t DataCounters(uint32_t* bytes_received, |
| uint32_t* packets_received) const; |
| |
| int32_t ResetStatistics(); |
| |
| int32_t ResetDataCounters(); |
| |
| uint16_t PacketOHReceived() const; |
| |
| uint32_t PacketCountReceived() const; |
| |
| uint32_t ByteCountReceived() const; |
| |
| int32_t RegisterRtpHeaderExtension(const RTPExtensionType type, |
| const uint8_t id); |
| |
| int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type); |
| |
| void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const; |
| |
| // RTX. |
| void SetRTXStatus(bool enable, uint32_t ssrc); |
| |
| void RTXStatus(bool* enable, uint32_t* ssrc, int* payload_type) const; |
| |
| void SetRtxPayloadType(int payload_type); |
| |
| virtual int8_t REDPayloadType() const; |
| |
| bool HaveNotReceivedPackets() const; |
| |
| virtual bool RetransmitOfOldPacket(const uint16_t sequence_number, |
| const uint32_t rtp_time_stamp) const; |
| |
| void UpdateStatistics(const WebRtcRTPHeader* rtp_header, |
| const uint16_t bytes, |
| const bool old_packet); |
| |
| private: |
| // Returns whether RED is configured with payload_type. |
| bool REDPayloadType(const int8_t payload_type) const; |
| |
| bool InOrderPacket(const uint16_t sequence_number) const; |
| |
| void CheckSSRCChanged(const WebRtcRTPHeader* rtp_header); |
| void CheckCSRC(const WebRtcRTPHeader* rtp_header); |
| int32_t CheckPayloadChanged(const WebRtcRTPHeader* rtp_header, |
| const int8_t first_payload_byte, |
| bool& isRED, |
| ModuleRTPUtility::PayloadUnion* payload); |
| |
| void UpdateNACKBitRate(int32_t bytes, uint32_t now); |
| bool ProcessNACKBitRate(uint32_t now); |
| |
| RTPPayloadRegistry* rtp_payload_registry_; |
| scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_; |
| |
| int32_t id_; |
| ModuleRtpRtcpImpl& rtp_rtcp_; |
| |
| RtpFeedback* cb_rtp_feedback_; |
| |
| CriticalSectionWrapper* critical_section_rtp_receiver_; |
| mutable int64_t last_receive_time_; |
| uint16_t last_received_payload_length_; |
| |
| uint32_t packet_timeout_ms_; |
| |
| RtpHeaderExtensionMap rtp_header_extension_map_; |
| |
| // SSRCs. |
| uint32_t ssrc_; |
| uint8_t num_csrcs_; |
| uint32_t current_remote_csrc_[kRtpCsrcSize]; |
| uint8_t num_energy_; |
| uint8_t current_remote_energy_[kRtpCsrcSize]; |
| |
| bool use_ssrc_filter_; |
| uint32_t ssrc_filter_; |
| |
| // Stats on received RTP packets. |
| uint32_t jitter_q4_; |
| mutable uint32_t jitter_max_q4_; |
| mutable uint32_t cumulative_loss_; |
| uint32_t jitter_q4_transmission_time_offset_; |
| |
| uint32_t local_time_last_received_timestamp_; |
| int64_t last_received_frame_time_ms_; |
| uint32_t last_received_timestamp_; |
| uint16_t last_received_sequence_number_; |
| int32_t last_received_transmission_time_offset_; |
| uint16_t received_seq_first_; |
| uint16_t received_seq_max_; |
| uint16_t received_seq_wraps_; |
| |
| // Current counter values. |
| uint16_t received_packet_oh_; |
| uint32_t received_byte_count_; |
| uint32_t received_old_packet_count_; |
| uint32_t received_inorder_packet_count_; |
| |
| // Counter values when we sent the last report. |
| mutable uint32_t last_report_inorder_packets_; |
| mutable uint32_t last_report_old_packets_; |
| mutable uint16_t last_report_seq_max_; |
| mutable uint8_t last_report_fraction_lost_; |
| mutable uint32_t last_report_cumulative_lost_; // 24 bits valid. |
| mutable uint32_t last_report_extended_high_seq_num_; |
| mutable uint32_t last_report_jitter_; |
| mutable uint32_t last_report_jitter_transmission_time_offset_; |
| |
| NACKMethod nack_method_; |
| int max_reordering_threshold_; |
| |
| bool rtx_; |
| uint32_t ssrc_rtx_; |
| int payload_type_rtx_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_ |