4ef438e2de
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
295 lines
11 KiB
C++
295 lines
11 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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// This sub-API supports the following functionalities:
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//
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// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
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// - SSRC handling.
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// - Transmission of RTCP sender reports.
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// - Obtaining RTCP data from incoming RTCP sender reports.
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// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
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// - Redundant Coding (RED)
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// - Writing RTP and RTCP packets to binary files for off-line analysis of
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// the call quality.
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//
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// Usage example, omitting error checking:
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//
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// using namespace webrtc;
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// VoiceEngine* voe = VoiceEngine::Create();
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// VoEBase* base = VoEBase::GetInterface(voe);
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// VoERTP_RTCP* rtp_rtcp = VoERTP_RTCP::GetInterface(voe);
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// base->Init();
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// int ch = base->CreateChannel();
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// ...
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// rtp_rtcp->SetLocalSSRC(ch, 12345);
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// ...
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// base->DeleteChannel(ch);
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// base->Terminate();
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// base->Release();
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// rtp_rtcp->Release();
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// VoiceEngine::Delete(voe);
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//
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#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
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#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
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#include <vector>
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#include "webrtc/common_types.h"
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namespace webrtc {
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class ViENetwork;
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class VoiceEngine;
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// VoERTPObserver
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class WEBRTC_DLLEXPORT VoERTPObserver
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{
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public:
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virtual void OnIncomingCSRCChanged(
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int channel, unsigned int CSRC, bool added) = 0;
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virtual void OnIncomingSSRCChanged(
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int channel, unsigned int SSRC) = 0;
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protected:
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virtual ~VoERTPObserver() {}
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};
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// VoERTCPObserver
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class WEBRTC_DLLEXPORT VoERTCPObserver
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{
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public:
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virtual void OnApplicationDataReceived(
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int channel, unsigned char subType,
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unsigned int name, const unsigned char* data,
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unsigned short dataLengthInBytes) = 0;
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protected:
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virtual ~VoERTCPObserver() {}
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};
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// CallStatistics
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struct CallStatistics
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{
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unsigned short fractionLost;
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unsigned int cumulativeLost;
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unsigned int extendedMax;
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unsigned int jitterSamples;
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int rttMs;
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int bytesSent;
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int packetsSent;
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int bytesReceived;
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int packetsReceived;
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// The capture ntp time (in local timebase) of the first played out audio
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// frame.
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int64_t capture_start_ntp_time_ms_;
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};
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// See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details.
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struct SenderInfo {
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uint32_t NTP_timestamp_high;
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uint32_t NTP_timestamp_low;
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uint32_t RTP_timestamp;
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uint32_t sender_packet_count;
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uint32_t sender_octet_count;
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};
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// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
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struct ReportBlock {
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uint32_t sender_SSRC; // SSRC of sender
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uint32_t source_SSRC;
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uint8_t fraction_lost;
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uint32_t cumulative_num_packets_lost;
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uint32_t extended_highest_sequence_number;
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uint32_t interarrival_jitter;
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uint32_t last_SR_timestamp;
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uint32_t delay_since_last_SR;
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};
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// VoERTP_RTCP
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class WEBRTC_DLLEXPORT VoERTP_RTCP
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{
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public:
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// Factory for the VoERTP_RTCP sub-API. Increases an internal
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// reference counter if successful. Returns NULL if the API is not
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// supported or if construction fails.
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static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
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// Releases the VoERTP_RTCP sub-API and decreases an internal
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// reference counter. Returns the new reference count. This value should
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// be zero for all sub-API:s before the VoiceEngine object can be safely
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// deleted.
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virtual int Release() = 0;
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// Sets the local RTP synchronization source identifier (SSRC) explicitly.
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virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
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// Gets the local RTP SSRC of a specified |channel|.
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virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
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// Gets the SSRC of the incoming RTP packets.
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virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
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// Sets the status of rtp-audio-level-indication on a specific |channel|.
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virtual int SetSendAudioLevelIndicationStatus(int channel,
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bool enable,
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unsigned char id = 1) = 0;
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// Sets the status of receiving rtp-audio-level-indication on a specific
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// |channel|.
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virtual int SetReceiveAudioLevelIndicationStatus(int channel,
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bool enable,
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unsigned char id = 1) {
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// TODO(wu): Remove default implementation once talk is updated.
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return 0;
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}
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// Sets the status of sending absolute sender time on a specific |channel|.
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virtual int SetSendAbsoluteSenderTimeStatus(int channel,
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bool enable,
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unsigned char id) = 0;
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// Sets status of receiving absolute sender time on a specific |channel|.
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virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
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bool enable,
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unsigned char id) = 0;
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// Sets the RTCP status on a specific |channel|.
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virtual int SetRTCPStatus(int channel, bool enable) = 0;
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// Gets the RTCP status on a specific |channel|.
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virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
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// Sets the canonical name (CNAME) parameter for RTCP reports on a
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// specific |channel|.
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virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
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// TODO(holmer): Remove this API once it has been removed from
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// fakewebrtcvoiceengine.h.
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virtual int GetRTCP_CNAME(int channel, char cName[256]) {
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return -1;
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}
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// Gets the canonical name (CNAME) parameter for incoming RTCP reports
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// on a specific channel.
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virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
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// Gets RTCP data from incoming RTCP Sender Reports.
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virtual int GetRemoteRTCPData(
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int channel, unsigned int& NTPHigh, unsigned int& NTPLow,
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unsigned int& timestamp, unsigned int& playoutTimestamp,
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unsigned int* jitter = NULL, unsigned short* fractionLost = NULL) = 0;
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// Gets RTP statistics for a specific |channel|.
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virtual int GetRTPStatistics(
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int channel, unsigned int& averageJitterMs, unsigned int& maxJitterMs,
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unsigned int& discardedPackets) = 0;
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// Gets RTCP statistics for a specific |channel|.
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virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
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// Gets the report block parts of the last received RTCP Sender Report (SR),
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// or RTCP Receiver Report (RR) on a specified |channel|. Each vector
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// element also contains the SSRC of the sender in addition to a report
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// block.
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virtual int GetRemoteRTCPReportBlocks(
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int channel, std::vector<ReportBlock>* receive_blocks) = 0;
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// Sets the Redundant Coding (RED) status on a specific |channel|.
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// TODO(minyue): Make SetREDStatus() pure virtual when fakewebrtcvoiceengine
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// in talk is ready.
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virtual int SetREDStatus(
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int channel, bool enable, int redPayloadtype = -1) { return -1; }
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// Gets the RED status on a specific |channel|.
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// TODO(minyue): Make GetREDStatus() pure virtual when fakewebrtcvoiceengine
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// in talk is ready.
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virtual int GetREDStatus(
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int channel, bool& enabled, int& redPayloadtype) { return -1; }
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// Sets the Forward Error Correction (FEC) status on a specific |channel|.
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// TODO(minyue): Remove SetFECStatus() when SetFECStatus() is replaced by
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// SetREDStatus() in fakewebrtcvoiceengine.
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virtual int SetFECStatus(
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int channel, bool enable, int redPayloadtype = -1) {
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return SetREDStatus(channel, enable, redPayloadtype);
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};
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// Gets the FEC status on a specific |channel|.
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// TODO(minyue): Remove GetFECStatus() when GetFECStatus() is replaced by
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// GetREDStatus() in fakewebrtcvoiceengine.
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virtual int GetFECStatus(
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int channel, bool& enabled, int& redPayloadtype) {
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return SetREDStatus(channel, enabled, redPayloadtype);
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}
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// This function enables Negative Acknowledgment (NACK) using RTCP,
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// implemented based on RFC 4585. NACK retransmits RTP packets if lost on
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// the network. This creates a lossless transport at the expense of delay.
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// If using NACK, NACK should be enabled on both endpoints in a call.
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virtual int SetNACKStatus(int channel,
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bool enable,
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int maxNoPackets) = 0;
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// Enables capturing of RTP packets to a binary file on a specific
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// |channel| and for a given |direction|. The file can later be replayed
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// using e.g. RTP Tools rtpplay since the binary file format is
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// compatible with the rtpdump format.
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virtual int StartRTPDump(
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int channel, const char fileNameUTF8[1024],
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RTPDirections direction = kRtpIncoming) = 0;
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// Disables capturing of RTP packets to a binary file on a specific
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// |channel| and for a given |direction|.
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virtual int StopRTPDump(
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int channel, RTPDirections direction = kRtpIncoming) = 0;
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// Gets the the current RTP capturing state for the specified
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// |channel| and |direction|.
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virtual int RTPDumpIsActive(
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int channel, RTPDirections direction = kRtpIncoming) = 0;
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// Sets video engine channel to receive incoming audio packets for
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// aggregated bandwidth estimation. Takes ownership of the ViENetwork
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// interface.
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virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network,
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int video_channel) {
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return 0;
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}
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// Will be removed. Don't use.
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virtual int RegisterRTPObserver(int channel,
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VoERTPObserver& observer) { return -1; };
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virtual int DeRegisterRTPObserver(int channel) { return -1; };
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virtual int RegisterRTCPObserver(
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int channel, VoERTCPObserver& observer) { return -1; };
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virtual int DeRegisterRTCPObserver(int channel) { return -1; };
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virtual int GetRemoteCSRCs(int channel,
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unsigned int arrCSRC[15]) { return -1; };
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virtual int InsertExtraRTPPacket(
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int channel, unsigned char payloadType, bool markerBit,
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const char* payloadData, unsigned short payloadSize) { return -1; };
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virtual int GetRemoteRTCPSenderInfo(
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int channel, SenderInfo* sender_info) { return -1; };
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virtual int SendApplicationDefinedRTCPPacket(
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int channel, unsigned char subType, unsigned int name,
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const char* data, unsigned short dataLengthInBytes) { return -1; };
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virtual int GetLastRemoteTimeStamp(int channel,
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uint32_t* lastRemoteTimeStamp) { return -1; };
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protected:
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VoERTP_RTCP() {}
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virtual ~VoERTP_RTCP() {}
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};
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} // namespace webrtc
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#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
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