244251a9cd
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video. BUG= TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test Review URL: https://webrtc-codereview.appspot.com/1078004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
244 lines
8.8 KiB
C++
244 lines
8.8 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 WebRtc_Word32 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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WebRtc_UWord32 MaxConfiguredBitrate() const;
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WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeout_ms);
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void PacketTimeout();
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void ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_Word64 now);
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void ProcessBitrate();
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WebRtc_Word32 RegisterReceivePayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payload_type,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payload_type);
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WebRtc_Word32 ReceivePayloadType(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate,
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WebRtc_Word8* payload_type) const;
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WebRtc_Word32 IncomingRTPPacket(
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WebRtcRTPHeader* rtpheader,
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const WebRtc_UWord8* incoming_rtp_packet,
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const WebRtc_UWord16 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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WebRtc_Word32 SetNACKStatus(const NACKMethod method,
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int max_reordering_threshold);
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// Returns the last received timestamp.
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virtual WebRtc_UWord32 TimeStamp() const;
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int32_t LastReceivedTimeMs() const;
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virtual WebRtc_UWord16 SequenceNumber() const;
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WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
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WebRtc_UWord32 SSRC() const;
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WebRtc_Word32 CSRCs(WebRtc_UWord32 array_of_csrc[kRtpCsrcSize]) const;
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WebRtc_Word32 Energy(WebRtc_UWord8 array_of_energy[kRtpCsrcSize]) const;
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// Get the currently configured SSRC filter.
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WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowed_ssrc) const;
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// Set a SSRC to be used as a filter for incoming RTP streams.
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WebRtc_Word32 SetSSRCFilter(const bool enable,
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const WebRtc_UWord32 allowed_ssrc);
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WebRtc_Word32 Statistics(WebRtc_UWord8* fraction_lost,
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WebRtc_UWord32* cum_lost,
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WebRtc_UWord32* ext_max,
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WebRtc_UWord32* jitter, // Will be moved from JB.
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WebRtc_UWord32* max_jitter,
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WebRtc_UWord32* jitter_transmission_time_offset,
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bool reset) const;
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WebRtc_Word32 Statistics(WebRtc_UWord8* fraction_lost,
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WebRtc_UWord32* cum_lost,
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WebRtc_UWord32* ext_max,
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WebRtc_UWord32* jitter, // Will be moved from JB.
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WebRtc_UWord32* max_jitter,
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WebRtc_UWord32* jitter_transmission_time_offset,
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WebRtc_Word32* missing,
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bool reset) const;
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WebRtc_Word32 DataCounters(WebRtc_UWord32* bytes_received,
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WebRtc_UWord32* packets_received) const;
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WebRtc_Word32 ResetStatistics();
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WebRtc_Word32 ResetDataCounters();
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WebRtc_UWord16 PacketOHReceived() const;
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WebRtc_UWord32 PacketCountReceived() const;
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WebRtc_UWord32 ByteCountReceived() const;
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WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
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const WebRtc_UWord8 id);
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WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
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void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const;
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// RTX.
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void SetRTXStatus(const bool enable, const WebRtc_UWord32 ssrc);
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void RTXStatus(bool* enable, WebRtc_UWord32* ssrc) const;
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virtual WebRtc_Word8 REDPayloadType() const;
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bool HaveNotReceivedPackets() const;
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virtual bool RetransmitOfOldPacket(const WebRtc_UWord16 sequence_number,
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const WebRtc_UWord32 rtp_time_stamp) const;
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void UpdateStatistics(const WebRtcRTPHeader* rtp_header,
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const WebRtc_UWord16 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 WebRtc_Word8 payload_type) const;
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bool InOrderPacket(const WebRtc_UWord16 sequence_number) const;
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void CheckSSRCChanged(const WebRtcRTPHeader* rtp_header);
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void CheckCSRC(const WebRtcRTPHeader* rtp_header);
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WebRtc_Word32 CheckPayloadChanged(const WebRtcRTPHeader* rtp_header,
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const WebRtc_Word8 first_payload_byte,
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bool& isRED,
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ModuleRTPUtility::PayloadUnion* payload);
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void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
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bool ProcessNACKBitRate(WebRtc_UWord32 now);
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RTPPayloadRegistry* rtp_payload_registry_;
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scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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WebRtc_Word32 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 WebRtc_Word64 last_receive_time_;
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WebRtc_UWord16 last_received_payload_length_;
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WebRtc_UWord32 packet_timeout_ms_;
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RtpHeaderExtensionMap rtp_header_extension_map_;
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// SSRCs.
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WebRtc_UWord32 ssrc_;
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WebRtc_UWord8 num_csrcs_;
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WebRtc_UWord32 current_remote_csrc_[kRtpCsrcSize];
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WebRtc_UWord8 num_energy_;
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WebRtc_UWord8 current_remote_energy_[kRtpCsrcSize];
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bool use_ssrc_filter_;
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WebRtc_UWord32 ssrc_filter_;
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// Stats on received RTP packets.
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WebRtc_UWord32 jitter_q4_;
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mutable WebRtc_UWord32 jitter_max_q4_;
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mutable WebRtc_UWord32 cumulative_loss_;
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WebRtc_UWord32 jitter_q4_transmission_time_offset_;
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WebRtc_UWord32 local_time_last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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WebRtc_UWord32 last_received_timestamp_;
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WebRtc_UWord16 last_received_sequence_number_;
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WebRtc_Word32 last_received_transmission_time_offset_;
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WebRtc_UWord16 received_seq_first_;
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WebRtc_UWord16 received_seq_max_;
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WebRtc_UWord16 received_seq_wraps_;
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// Current counter values.
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WebRtc_UWord16 received_packet_oh_;
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WebRtc_UWord32 received_byte_count_;
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WebRtc_UWord32 received_old_packet_count_;
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WebRtc_UWord32 received_inorder_packet_count_;
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// Counter values when we sent the last report.
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mutable WebRtc_UWord32 last_report_inorder_packets_;
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mutable WebRtc_UWord32 last_report_old_packets_;
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mutable WebRtc_UWord16 last_report_seq_max_;
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mutable WebRtc_UWord8 last_report_fraction_lost_;
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mutable WebRtc_UWord32 last_report_cumulative_lost_; // 24 bits valid.
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mutable WebRtc_UWord32 last_report_extended_high_seq_num_;
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mutable WebRtc_UWord32 last_report_jitter_;
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mutable WebRtc_UWord32 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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WebRtc_UWord32 ssrc_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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