7c4d20fd6c
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls by doing a lot shorter locking which fetches a current state of RTPSenderAudio variables before sending. Thread annotates locked variables and removes one lock in RTPSenderAudio, bonus fixes data races reported in voe_auto_test --automated under TSan (DTMF data race). Also includes some bonus cleanup of RTPSenderVideo which removes the send critsect completely as all methods using it was always called from RTPSender under its send_critsect. R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org BUG=3001, chromium:454654 Review URL: https://webrtc-codereview.appspot.com/41869004 Cr-Commit-Position: refs/heads/master@{#8348} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
425 lines
14 KiB
C++
425 lines
14 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_RTCP_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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#include <list>
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#include <vector>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/test/testsupport/gtest_prod_util.h"
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namespace webrtc {
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class ModuleRtpRtcpImpl : public RtpRtcp {
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public:
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explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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virtual ~ModuleRtpRtcpImpl();
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// Returns the number of milliseconds until the module want a worker thread to
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// call Process.
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virtual int64_t TimeUntilNextProcess() OVERRIDE;
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// Process any pending tasks such as timeouts.
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virtual int32_t Process() OVERRIDE;
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// Receiver part.
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// Called when we receive an RTCP packet.
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virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
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size_t incoming_packet_length) OVERRIDE;
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virtual void SetRemoteSSRC(uint32_t ssrc) OVERRIDE;
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// Sender part.
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virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) OVERRIDE;
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virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) OVERRIDE;
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virtual int32_t DeRegisterSendPayload(int8_t payload_type) OVERRIDE;
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int8_t SendPayloadType() const;
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// Register RTP header extension.
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virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) OVERRIDE;
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virtual int32_t DeregisterSendRtpHeaderExtension(
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RTPExtensionType type) OVERRIDE;
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// Get start timestamp.
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virtual uint32_t StartTimestamp() const OVERRIDE;
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// Configure start timestamp, default is a random number.
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virtual void SetStartTimestamp(uint32_t timestamp) OVERRIDE;
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virtual uint16_t SequenceNumber() const OVERRIDE;
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// Set SequenceNumber, default is a random number.
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virtual void SetSequenceNumber(uint16_t seq) OVERRIDE;
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virtual void SetRtpStateForSsrc(uint32_t ssrc,
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const RtpState& rtp_state) OVERRIDE;
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virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) OVERRIDE;
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virtual uint32_t SSRC() const OVERRIDE;
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// Configure SSRC, default is a random number.
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virtual void SetSSRC(uint32_t ssrc) OVERRIDE;
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virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) OVERRIDE;
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RTCPSender::FeedbackState GetFeedbackState();
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int CurrentSendFrequencyHz() const;
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virtual void SetRtxSendStatus(int mode) OVERRIDE;
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virtual int RtxSendStatus() const OVERRIDE;
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virtual void SetRtxSsrc(uint32_t ssrc) OVERRIDE;
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virtual void SetRtxSendPayloadType(int payload_type) OVERRIDE;
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// Sends kRtcpByeCode when going from true to false.
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virtual int32_t SetSendingStatus(bool sending) OVERRIDE;
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virtual bool Sending() const OVERRIDE;
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// Drops or relays media packets.
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virtual void SetSendingMediaStatus(bool sending) OVERRIDE;
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virtual bool SendingMedia() const OVERRIDE;
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// Used by the codec module to deliver a video or audio frame for
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// packetization.
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virtual int32_t SendOutgoingData(
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FrameType frame_type,
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int8_t payload_type,
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uint32_t time_stamp,
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int64_t capture_time_ms,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtp_video_hdr = NULL) OVERRIDE;
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virtual bool TimeToSendPacket(uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_time_ms,
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bool retransmission) OVERRIDE;
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// Returns the number of padding bytes actually sent, which can be more or
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// less than |bytes|.
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virtual size_t TimeToSendPadding(size_t bytes) OVERRIDE;
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virtual bool GetSendSideDelay(int* avg_send_delay_ms,
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int* max_send_delay_ms) const OVERRIDE;
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// RTCP part.
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// Get RTCP status.
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virtual RTCPMethod RTCP() const OVERRIDE;
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// Configure RTCP status i.e on/off.
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virtual void SetRTCPStatus(RTCPMethod method) OVERRIDE;
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// Set RTCP CName.
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virtual int32_t SetCNAME(const char c_name[RTCP_CNAME_SIZE]) OVERRIDE;
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// Get remote CName.
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virtual int32_t RemoteCNAME(uint32_t remote_ssrc,
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char c_name[RTCP_CNAME_SIZE]) const OVERRIDE;
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// Get remote NTP.
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virtual int32_t RemoteNTP(uint32_t* received_ntp_secs,
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uint32_t* received_ntp_frac,
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uint32_t* rtcp_arrival_time_secs,
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uint32_t* rtcp_arrival_time_frac,
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uint32_t* rtcp_timestamp) const OVERRIDE;
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virtual int32_t AddMixedCNAME(uint32_t ssrc,
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const char c_name[RTCP_CNAME_SIZE]) OVERRIDE;
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virtual int32_t RemoveMixedCNAME(uint32_t ssrc) OVERRIDE;
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// Get RoundTripTime.
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virtual int32_t RTT(uint32_t remote_ssrc,
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int64_t* rtt,
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int64_t* avg_rtt,
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int64_t* min_rtt,
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int64_t* max_rtt) const OVERRIDE;
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// Force a send of an RTCP packet.
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// Normal SR and RR are triggered via the process function.
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virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) OVERRIDE;
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virtual int32_t ResetSendDataCountersRTP() OVERRIDE;
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// Statistics of the amount of data sent and received.
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virtual int32_t DataCountersRTP(size_t* bytes_sent,
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uint32_t* packets_sent) const OVERRIDE;
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virtual void GetSendStreamDataCounters(
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StreamDataCounters* rtp_counters,
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StreamDataCounters* rtx_counters) const OVERRIDE;
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// Get received RTCP report, sender info.
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virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) OVERRIDE;
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// Get received RTCP report, report block.
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virtual int32_t RemoteRTCPStat(
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std::vector<RTCPReportBlock>* receive_blocks) const OVERRIDE;
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// Set received RTCP report block.
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virtual int32_t AddRTCPReportBlock(
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uint32_t ssrc,
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const RTCPReportBlock* receive_block) OVERRIDE;
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virtual int32_t RemoveRTCPReportBlock(uint32_t ssrc) OVERRIDE;
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virtual void GetRtcpPacketTypeCounters(
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RtcpPacketTypeCounter* packets_sent,
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RtcpPacketTypeCounter* packets_received) const OVERRIDE;
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// (REMB) Receiver Estimated Max Bitrate.
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virtual bool REMB() const OVERRIDE;
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virtual void SetREMBStatus(bool enable) OVERRIDE;
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virtual void SetREMBData(uint32_t bitrate,
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const std::vector<uint32_t>& ssrcs) OVERRIDE;
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// (IJ) Extended jitter report.
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virtual bool IJ() const OVERRIDE;
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virtual void SetIJStatus(bool enable) OVERRIDE;
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// (TMMBR) Temporary Max Media Bit Rate.
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virtual bool TMMBR() const OVERRIDE;
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virtual void SetTMMBRStatus(bool enable) OVERRIDE;
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int32_t SetTMMBN(const TMMBRSet* bounding_set);
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virtual uint16_t MaxPayloadLength() const OVERRIDE;
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virtual uint16_t MaxDataPayloadLength() const OVERRIDE;
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virtual int32_t SetMaxTransferUnit(uint16_t size) OVERRIDE;
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virtual int32_t SetTransportOverhead(
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bool tcp,
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bool ipv6,
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uint8_t authentication_overhead = 0) OVERRIDE;
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// (NACK) Negative acknowledgment part.
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virtual int SelectiveRetransmissions() const OVERRIDE;
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virtual int SetSelectiveRetransmissions(uint8_t settings) OVERRIDE;
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// Send a Negative acknowledgment packet.
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virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) OVERRIDE;
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// Store the sent packets, needed to answer to a negative acknowledgment
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// requests.
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virtual void SetStorePacketsStatus(bool enable,
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uint16_t number_to_store) OVERRIDE;
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virtual bool StorePackets() const OVERRIDE;
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// Called on receipt of RTCP report block from remote side.
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virtual void RegisterRtcpStatisticsCallback(
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RtcpStatisticsCallback* callback) OVERRIDE;
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virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() OVERRIDE;
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// (APP) Application specific data.
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virtual int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
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uint32_t name,
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const uint8_t* data,
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uint16_t length) OVERRIDE;
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// (XR) VOIP metric.
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virtual int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) OVERRIDE;
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// (XR) Receiver reference time report.
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virtual void SetRtcpXrRrtrStatus(bool enable) OVERRIDE;
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virtual bool RtcpXrRrtrStatus() const OVERRIDE;
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// Audio part.
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// Set audio packet size, used to determine when it's time to send a DTMF
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// packet in silence (CNG).
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virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) OVERRIDE;
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// Send a TelephoneEvent tone using RFC 2833 (4733).
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virtual int32_t SendTelephoneEventOutband(uint8_t key,
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uint16_t time_ms,
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uint8_t level) OVERRIDE;
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// Set payload type for Redundant Audio Data RFC 2198.
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virtual int32_t SetSendREDPayloadType(int8_t payload_type) OVERRIDE;
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// Get payload type for Redundant Audio Data RFC 2198.
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virtual int32_t SendREDPayloadType(int8_t& payload_type) const OVERRIDE;
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// Store the audio level in d_bov for header-extension-for-audio-level-
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// indication.
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virtual int32_t SetAudioLevel(uint8_t level_d_bov) OVERRIDE;
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// Video part.
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virtual int32_t SendRTCPSliceLossIndication(uint8_t picture_id) OVERRIDE;
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// Set method for requesting a new key frame.
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virtual int32_t SetKeyFrameRequestMethod(
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KeyFrameRequestMethod method) OVERRIDE;
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// Send a request for a keyframe.
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virtual int32_t RequestKeyFrame() OVERRIDE;
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virtual int32_t SetCameraDelay(int32_t delay_ms) OVERRIDE;
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virtual void SetTargetSendBitrate(
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const std::vector<uint32_t>& stream_bitrates) OVERRIDE;
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virtual int32_t SetGenericFECStatus(bool enable,
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uint8_t payload_type_red,
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uint8_t payload_type_fec) OVERRIDE;
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virtual int32_t GenericFECStatus(
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bool& enable,
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uint8_t& payload_type_red,
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uint8_t& payload_type_fec) OVERRIDE;
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virtual int32_t SetFecParameters(
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const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params) OVERRIDE;
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bool LastReceivedNTP(uint32_t* NTPsecs,
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uint32_t* NTPfrac,
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uint32_t* remote_sr) const;
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bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
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virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
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virtual void BitrateSent(uint32_t* total_rate,
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uint32_t* video_rate,
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uint32_t* fec_rate,
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uint32_t* nackRate) const OVERRIDE;
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int64_t SendTimeOfSendReport(uint32_t send_report);
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bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
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// Good state of RTP receiver inform sender.
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virtual int32_t SendRTCPReferencePictureSelection(
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uint64_t picture_id) OVERRIDE;
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virtual void RegisterSendChannelRtpStatisticsCallback(
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StreamDataCountersCallback* callback) OVERRIDE;
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virtual StreamDataCountersCallback*
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GetSendChannelRtpStatisticsCallback() const OVERRIDE;
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void OnReceivedTMMBR();
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// Bad state of RTP receiver request a keyframe.
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void OnRequestIntraFrame();
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// Received a request for a new SLI.
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void OnReceivedSliceLossIndication(uint8_t picture_id);
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// Received a new reference frame.
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void OnReceivedReferencePictureSelectionIndication(uint64_t picture_id);
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void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
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void OnRequestSendReport();
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protected:
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void RegisterChildModule(RtpRtcp* module);
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void DeRegisterChildModule(RtpRtcp* module);
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bool UpdateRTCPReceiveInformationTimers();
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uint32_t BitrateReceivedNow() const;
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// Get remote SequenceNumber.
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uint16_t RemoteSequenceNumber() const;
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// Only for internal testing.
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uint32_t LastSendReport(int64_t& last_rtcptime);
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RTPSender rtp_sender_;
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RTCPSender rtcp_sender_;
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RTCPReceiver rtcp_receiver_;
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Clock* clock_;
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private:
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FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
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FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
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int64_t RtcpReportInterval();
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void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
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void set_rtt_ms(int64_t rtt_ms);
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int64_t rtt_ms() const;
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bool TimeToSendFullNackList(int64_t now) const;
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bool IsDefaultModule() const;
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int32_t id_;
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const bool audio_;
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bool collision_detected_;
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int64_t last_process_time_;
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int64_t last_bitrate_process_time_;
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int64_t last_rtt_process_time_;
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uint16_t packet_overhead_;
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scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
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scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_;
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ModuleRtpRtcpImpl* default_module_;
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std::vector<ModuleRtpRtcpImpl*> child_modules_;
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size_t padding_index_;
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// Send side
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NACKMethod nack_method_;
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int64_t nack_last_time_sent_full_;
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uint32_t nack_last_time_sent_full_prev_;
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uint16_t nack_last_seq_number_sent_;
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bool simulcast_;
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VideoCodec send_video_codec_;
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KeyFrameRequestMethod key_frame_req_method_;
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RemoteBitrateEstimator* remote_bitrate_;
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RtcpRttStats* rtt_stats_;
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// The processed RTT from RtcpRttStats.
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scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
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int64_t rtt_ms_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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