a5cb98cbbd
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
243 lines
8.0 KiB
C++
243 lines
8.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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#include <map>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RtpRtcpFeedback;
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class ModuleRtpRtcpImpl;
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class Trace;
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class RTPReceiverAudio;
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class RTPReceiverVideo;
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class RTPReceiverStrategy;
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class RTPReceiver : public Bitrate {
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public:
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// Callbacks passed in here may not be NULL (use Null Object callbacks if you
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// want callbacks to do nothing). This class takes ownership of the media
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// receiver but nothing else.
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RTPReceiver(const int32_t id,
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Clock* clock,
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ModuleRtpRtcpImpl* owner,
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RtpAudioFeedback* incoming_audio_messages_callback,
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RtpData* incoming_payload_callback,
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RtpFeedback* incoming_messages_callback,
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RTPReceiverStrategy* rtp_media_receiver,
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RTPPayloadRegistry* rtp_payload_registry);
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virtual ~RTPReceiver();
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RtpVideoCodecTypes VideoCodecType() const;
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uint32_t MaxConfiguredBitrate() const;
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int32_t SetPacketTimeout(const uint32_t timeout_ms);
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void PacketTimeout();
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void ProcessDeadOrAlive(const bool RTCPalive, const int64_t now);
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void ProcessBitrate();
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int32_t RegisterReceivePayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate);
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int32_t DeRegisterReceivePayload(const int8_t payload_type);
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int32_t ReceivePayloadType(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate,
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int8_t* payload_type) const;
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int32_t IncomingRTPPacket(
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RTPHeader* rtpheader,
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const uint8_t* incoming_rtp_packet,
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const uint16_t incoming_rtp_packet_length);
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NACKMethod NACK() const ;
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// Turn negative acknowledgement requests on/off.
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int32_t SetNACKStatus(const NACKMethod method, int max_reordering_threshold);
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// Returns the last received timestamp.
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virtual uint32_t TimeStamp() const;
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int32_t LastReceivedTimeMs() const;
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virtual uint16_t SequenceNumber() const;
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int32_t EstimatedRemoteTimeStamp(uint32_t& timestamp) const;
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uint32_t SSRC() const;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const;
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int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
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// Get the currently configured SSRC filter.
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int32_t SSRCFilter(uint32_t& allowed_ssrc) const;
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// Set a SSRC to be used as a filter for incoming RTP streams.
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int32_t SetSSRCFilter(const bool enable, const uint32_t allowed_ssrc);
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int32_t Statistics(uint8_t* fraction_lost,
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uint32_t* cum_lost,
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uint32_t* ext_max,
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uint32_t* jitter, // Will be moved from JB.
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uint32_t* max_jitter,
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uint32_t* jitter_transmission_time_offset,
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bool reset) const;
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int32_t Statistics(uint8_t* fraction_lost,
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uint32_t* cum_lost,
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uint32_t* ext_max,
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uint32_t* jitter, // Will be moved from JB.
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uint32_t* max_jitter,
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uint32_t* jitter_transmission_time_offset,
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int32_t* missing,
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bool reset) const;
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int32_t DataCounters(uint32_t* bytes_received,
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uint32_t* packets_received) const;
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int32_t ResetStatistics();
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int32_t ResetDataCounters();
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uint16_t PacketOHReceived() const;
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uint32_t PacketCountReceived() const;
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uint32_t ByteCountReceived() const;
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int32_t RegisterRtpHeaderExtension(const RTPExtensionType type,
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const uint8_t id);
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int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type);
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void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const;
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// RTX.
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void SetRTXStatus(bool enable, uint32_t ssrc);
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void RTXStatus(bool* enable, uint32_t* ssrc, int* payload_type) const;
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void SetRtxPayloadType(int payload_type);
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virtual int8_t REDPayloadType() const;
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bool HaveNotReceivedPackets() const;
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virtual bool RetransmitOfOldPacket(const uint16_t sequence_number,
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const uint32_t rtp_time_stamp) const;
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void UpdateStatistics(const RTPHeader* rtp_header,
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const uint16_t bytes,
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const bool old_packet);
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private:
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// Returns whether RED is configured with payload_type.
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bool REDPayloadType(const int8_t payload_type) const;
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bool InOrderPacket(const uint16_t sequence_number) const;
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void CheckSSRCChanged(const RTPHeader* rtp_header);
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void CheckCSRC(const WebRtcRTPHeader* rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader* rtp_header,
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const int8_t first_payload_byte,
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bool& isRED,
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ModuleRTPUtility::PayloadUnion* payload);
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void UpdateNACKBitRate(int32_t bytes, uint32_t now);
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bool ProcessNACKBitRate(uint32_t now);
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RTPPayloadRegistry* rtp_payload_registry_;
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scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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int32_t id_;
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ModuleRtpRtcpImpl& rtp_rtcp_;
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RtpFeedback* cb_rtp_feedback_;
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CriticalSectionWrapper* critical_section_rtp_receiver_;
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mutable int64_t last_receive_time_;
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uint16_t last_received_payload_length_;
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uint32_t packet_timeout_ms_;
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// SSRCs.
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uint32_t ssrc_;
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uint8_t num_csrcs_;
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uint32_t current_remote_csrc_[kRtpCsrcSize];
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uint8_t num_energy_;
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uint8_t current_remote_energy_[kRtpCsrcSize];
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bool use_ssrc_filter_;
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uint32_t ssrc_filter_;
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// Stats on received RTP packets.
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uint32_t jitter_q4_;
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mutable uint32_t jitter_max_q4_;
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mutable uint32_t cumulative_loss_;
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uint32_t jitter_q4_transmission_time_offset_;
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uint32_t local_time_last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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uint32_t last_received_timestamp_;
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uint16_t last_received_sequence_number_;
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int32_t last_received_transmission_time_offset_;
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uint16_t received_seq_first_;
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uint16_t received_seq_max_;
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uint16_t received_seq_wraps_;
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// Current counter values.
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uint16_t received_packet_oh_;
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uint32_t received_byte_count_;
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uint32_t received_old_packet_count_;
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uint32_t received_inorder_packet_count_;
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// Counter values when we sent the last report.
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mutable uint32_t last_report_inorder_packets_;
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mutable uint32_t last_report_old_packets_;
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mutable uint16_t last_report_seq_max_;
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mutable uint8_t last_report_fraction_lost_;
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mutable uint32_t last_report_cumulative_lost_; // 24 bits valid.
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mutable uint32_t last_report_extended_high_seq_num_;
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mutable uint32_t last_report_jitter_;
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mutable uint32_t last_report_jitter_transmission_time_offset_;
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NACKMethod nack_method_;
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int max_reordering_threshold_;
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bool rtx_;
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uint32_t ssrc_rtx_;
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int payload_type_rtx_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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