efae5d5901
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine. BUG= TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test Review URL: https://webrtc-codereview.appspot.com/1022011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
258 lines
9.2 KiB
C++
258 lines
9.2 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).
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RTPReceiver(const WebRtc_Word32 id,
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const bool audio,
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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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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 ReceivePayload(const WebRtc_Word8 payload_type,
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char payload_name[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels,
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WebRtc_UWord32* rate) 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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// 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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virtual WebRtc_UWord32 PayloadTypeToPayload(
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const WebRtc_UWord8 payload_type,
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ModuleRTPUtility::Payload*& payload) 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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RTPReceiverAudio* GetAudioReceiver() const {
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return rtp_receiver_audio_.get();
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}
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virtual WebRtc_Word8 REDPayloadType() const;
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bool HaveNotReceivedPackets() const;
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protected:
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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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private:
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RTPPayloadRegistry rtp_payload_registry_;
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scoped_ptr<RTPReceiverAudio> rtp_receiver_audio_;
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scoped_ptr<RTPReceiverVideo> rtp_receiver_video_;
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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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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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