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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This sub-API supports the following functionalities:
//
// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
// - SSRC handling.
// - Transmission of RTCP sender reports.
// - Obtaining RTCP data from incoming RTCP sender reports.
// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
// - Redundant Coding (RED)
// - Writing RTP and RTCP packets to binary files for off-line analysis of
// the call quality.
//
// Usage example, omitting error checking:
//
// using namespace webrtc;
// VoiceEngine* voe = VoiceEngine::Create();
// VoEBase* base = VoEBase::GetInterface(voe);
// VoERTP_RTCP* rtp_rtcp = VoERTP_RTCP::GetInterface(voe);
// base->Init();
// int ch = base->CreateChannel();
// ...
// rtp_rtcp->SetLocalSSRC(ch, 12345);
// ...
// base->DeleteChannel(ch);
// base->Terminate();
// base->Release();
// rtp_rtcp->Release();
// VoiceEngine::Delete(voe);
//
#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
#include <vector>
#include "webrtc/common_types.h"
namespace webrtc {
class VoiceEngine;
// VoERTPObserver
class WEBRTC_DLLEXPORT VoERTPObserver {
public:
virtual void OnIncomingCSRCChanged(int channel,
unsigned int CSRC,
bool added) = 0;
virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0;
protected:
virtual ~VoERTPObserver() {}
};
// CallStatistics
struct CallStatistics {
unsigned short fractionLost;
unsigned int cumulativeLost;
unsigned int extendedMax;
unsigned int jitterSamples;
int64_t rttMs;
size_t bytesSent;
int packetsSent;
size_t bytesReceived;
int packetsReceived;
// The capture ntp time (in local timebase) of the first played out audio
// frame.
int64_t capture_start_ntp_time_ms_;
};
// See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details.
struct SenderInfo {
uint32_t NTP_timestamp_high;
uint32_t NTP_timestamp_low;
uint32_t RTP_timestamp;
uint32_t sender_packet_count;
uint32_t sender_octet_count;
};
// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
struct ReportBlock {
uint32_t sender_SSRC; // SSRC of sender
uint32_t source_SSRC;
uint8_t fraction_lost;
uint32_t cumulative_num_packets_lost;
uint32_t extended_highest_sequence_number;
uint32_t interarrival_jitter;
uint32_t last_SR_timestamp;
uint32_t delay_since_last_SR;
};
// VoERTP_RTCP
class WEBRTC_DLLEXPORT VoERTP_RTCP {
public:
// Factory for the VoERTP_RTCP sub-API. Increases an internal
// reference counter if successful. Returns NULL if the API is not
// supported or if construction fails.
static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
// Releases the VoERTP_RTCP sub-API and decreases an internal
// reference counter. Returns the new reference count. This value should
// be zero for all sub-API:s before the VoiceEngine object can be safely
// deleted.
virtual int Release() = 0;
// Sets the local RTP synchronization source identifier (SSRC) explicitly.
virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
// Gets the local RTP SSRC of a specified |channel|.
virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
// Gets the SSRC of the incoming RTP packets.
virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
// Sets the status of rtp-audio-level-indication on a specific |channel|.
virtual int SetSendAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id = 1) = 0;
// Sets the status of receiving rtp-audio-level-indication on a specific
// |channel|.
virtual int SetReceiveAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id = 1) {
// TODO(wu): Remove default implementation once talk is updated.
return 0;
}
// Sets the status of sending absolute sender time on a specific |channel|.
virtual int SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) = 0;
// Sets status of receiving absolute sender time on a specific |channel|.
virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) = 0;
// Sets the RTCP status on a specific |channel|.
virtual int SetRTCPStatus(int channel, bool enable) = 0;
// Gets the RTCP status on a specific |channel|.
virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
// Sets the canonical name (CNAME) parameter for RTCP reports on a
// specific |channel|.
virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
// TODO(holmer): Remove this API once it has been removed from
// fakewebrtcvoiceengine.h.
virtual int GetRTCP_CNAME(int channel, char cName[256]) { return -1; }
// Gets the canonical name (CNAME) parameter for incoming RTCP reports
// on a specific channel.
virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
// Gets RTCP data from incoming RTCP Sender Reports.
virtual int GetRemoteRTCPData(int channel,
unsigned int& NTPHigh,
unsigned int& NTPLow,
unsigned int& timestamp,
unsigned int& playoutTimestamp,
unsigned int* jitter = NULL,
unsigned short* fractionLost = NULL) = 0;
// Gets RTP statistics for a specific |channel|.
virtual int GetRTPStatistics(int channel,
unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets) = 0;
// Gets RTCP statistics for a specific |channel|.
virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
// Gets the report block parts of the last received RTCP Sender Report (SR),
// or RTCP Receiver Report (RR) on a specified |channel|. Each vector
// element also contains the SSRC of the sender in addition to a report
// block.
virtual int GetRemoteRTCPReportBlocks(
int channel,
std::vector<ReportBlock>* receive_blocks) = 0;
// Sets the Redundant Coding (RED) status on a specific |channel|.
// TODO(minyue): Make SetREDStatus() pure virtual when fakewebrtcvoiceengine
// in talk is ready.
virtual int SetREDStatus(int channel, bool enable, int redPayloadtype = -1) {
return -1;
}
// Gets the RED status on a specific |channel|.
// TODO(minyue): Make GetREDStatus() pure virtual when fakewebrtcvoiceengine
// in talk is ready.
virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype) {
return -1;
}
// Sets the Forward Error Correction (FEC) status on a specific |channel|.
// TODO(minyue): Remove SetFECStatus() when SetFECStatus() is replaced by
// SetREDStatus() in fakewebrtcvoiceengine.
virtual int SetFECStatus(int channel, bool enable, int redPayloadtype = -1) {
return SetREDStatus(channel, enable, redPayloadtype);
};
// Gets the FEC status on a specific |channel|.
// TODO(minyue): Remove GetFECStatus() when GetFECStatus() is replaced by
// GetREDStatus() in fakewebrtcvoiceengine.
virtual int GetFECStatus(int channel, bool& enabled, int& redPayloadtype) {
return SetREDStatus(channel, enabled, redPayloadtype);
}
// This function enables Negative Acknowledgment (NACK) using RTCP,
// implemented based on RFC 4585. NACK retransmits RTP packets if lost on
// the network. This creates a lossless transport at the expense of delay.
// If using NACK, NACK should be enabled on both endpoints in a call.
virtual int SetNACKStatus(int channel, bool enable, int maxNoPackets) = 0;
protected:
VoERTP_RTCP() {}
virtual ~VoERTP_RTCP() {}
};
} // namespace webrtc
#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
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