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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);
#include <vector>
#include "webrtc/common_types.h"
namespace webrtc {
class VoiceEngine;
// VoERTPObserver
virtual void OnIncomingCSRCChanged(int channel,
unsigned int CSRC,
bool added) = 0;
virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0;
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 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 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;
// 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 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;
// 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 statistics for a specific |channel|.
virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
virtual ~VoERTP_RTCP() {}
} // namespace webrtc