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Remove VoERTP_RTCP.

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BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3006383002
Cr-Commit-Position: refs/heads/master@{#19892}
This commit is contained in:
solenberg 2017-09-18 07:05:30 -07:00 committed by Commit Bot
parent c5267d251a
commit dd3abbb532
10 changed files with 30 additions and 547 deletions
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38
test/mock_voice_engine.h
View File

@ -127,44 +127,6 @@ class MockVoiceEngine : public VoiceEngineImpl {
MOCK_METHOD3(ReceivedRTCPPacket,
int(int channel, const void* data, size_t length));
// VoERTP_RTCP
MOCK_METHOD2(SetLocalSSRC, int(int channel, unsigned int ssrc));
MOCK_METHOD2(GetLocalSSRC, int(int channel, unsigned int& ssrc));
MOCK_METHOD2(GetRemoteSSRC, int(int channel, unsigned int& ssrc));
MOCK_METHOD3(SetSendAudioLevelIndicationStatus,
int(int channel, bool enable, unsigned char id));
MOCK_METHOD3(SetReceiveAudioLevelIndicationStatus,
int(int channel, bool enable, unsigned char id));
MOCK_METHOD3(SetSendAbsoluteSenderTimeStatus,
int(int channel, bool enable, unsigned char id));
MOCK_METHOD3(SetReceiveAbsoluteSenderTimeStatus,
int(int channel, bool enable, unsigned char id));
MOCK_METHOD2(SetRTCPStatus, int(int channel, bool enable));
MOCK_METHOD2(GetRTCPStatus, int(int channel, bool& enabled));
MOCK_METHOD2(SetRTCP_CNAME, int(int channel, const char cName[256]));
MOCK_METHOD2(GetRTCP_CNAME, int(int channel, char cName[256]));
MOCK_METHOD2(GetRemoteRTCP_CNAME, int(int channel, char cName[256]));
MOCK_METHOD7(GetRemoteRTCPData,
int(int channel,
unsigned int& NTPHigh,
unsigned int& NTPLow,
unsigned int& timestamp,
unsigned int& playoutTimestamp,
unsigned int* jitter,
unsigned short* fraction_lost));
MOCK_METHOD4(GetRTPStatistics,
int(int channel,
unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets));
MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics& stats));
MOCK_METHOD2(GetRemoteRTCPReportBlocks,
int(int channel, std::vector<ReportBlock>* receive_blocks));
MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype));
MOCK_METHOD3(GetREDStatus,
int(int channel, bool& enable, int& redPayloadtype));
MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets));
private:
// TODO(ossu): I'm not particularly happy about keeping the decoder factory
// here, but due to how gmock is implemented, I cannot just keep it in the

3
voice_engine/BUILD.gn
View File

@ -19,7 +19,6 @@ rtc_static_library("voice_engine") {
"include/voe_base.h",
"include/voe_errors.h",
"include/voe_network.h",
"include/voe_rtp_rtcp.h",
"monitor_module.h",
"output_mixer.cc",
"output_mixer.h",
@ -37,8 +36,6 @@ rtc_static_library("voice_engine") {
"voe_base_impl.h",
"voe_network_impl.cc",
"voe_network_impl.h",
"voe_rtp_rtcp_impl.cc",
"voe_rtp_rtcp_impl.h",
"voice_engine_defines.h",
"voice_engine_impl.cc",
"voice_engine_impl.h",

75
voice_engine/channel.cc
View File

@ -39,7 +39,6 @@
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/include/voe_rtp_rtcp.h"
#include "voice_engine/output_mixer.h"
#include "voice_engine/statistics.h"
#include "voice_engine/utility.h"
@ -1576,11 +1575,6 @@ int Channel::SetLocalSSRC(unsigned int ssrc) {
return 0;
}
int Channel::GetLocalSSRC(unsigned int& ssrc) {
ssrc = _rtpRtcpModule->SSRC();
return 0;
}
int Channel::GetRemoteSSRC(unsigned int& ssrc) {
ssrc = rtp_receiver_->SSRC();
return 0;
@ -1671,12 +1665,6 @@ void Channel::SetRTCPStatus(bool enable) {
_rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
}
int Channel::GetRTCPStatus(bool& enabled) {
RtcpMode method = _rtpRtcpModule->RTCP();
enabled = (method != RtcpMode::kOff);
return 0;
}
int Channel::SetRTCP_CNAME(const char cName[256]) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCP_CNAME()");
@ -1689,69 +1677,6 @@ int Channel::SetRTCP_CNAME(const char cName[256]) {
return 0;
}
int Channel::GetRemoteRTCP_CNAME(char cName[256]) {
if (cName == NULL) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"GetRemoteRTCP_CNAME() invalid CNAME input buffer");
return -1;
}
char cname[RTCP_CNAME_SIZE];
const uint32_t remoteSSRC = rtp_receiver_->SSRC();
if (_rtpRtcpModule->RemoteCNAME(remoteSSRC, cname) != 0) {
_engineStatisticsPtr->SetLastError(
VE_CANNOT_RETRIEVE_CNAME, kTraceError,
"GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME");
return -1;
}
strcpy(cName, cname);
return 0;
}
int Channel::SendApplicationDefinedRTCPPacket(
unsigned char subType,
unsigned int name,
const char* data,
unsigned short dataLengthInBytes) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendApplicationDefinedRTCPPacket()");
if (!channel_state_.Get().sending) {
_engineStatisticsPtr->SetLastError(
VE_NOT_SENDING, kTraceError,
"SendApplicationDefinedRTCPPacket() not sending");
return -1;
}
if (NULL == data) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SendApplicationDefinedRTCPPacket() invalid data value");
return -1;
}
if (dataLengthInBytes % 4 != 0) {
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SendApplicationDefinedRTCPPacket() invalid length value");
return -1;
}
RtcpMode status = _rtpRtcpModule->RTCP();
if (status == RtcpMode::kOff) {
_engineStatisticsPtr->SetLastError(
VE_RTCP_ERROR, kTraceError,
"SendApplicationDefinedRTCPPacket() RTCP is disabled");
return -1;
}
// Create and schedule the RTCP APP packet for transmission
if (_rtpRtcpModule->SetRTCPApplicationSpecificData(
subType, name, (const unsigned char*)data, dataLengthInBytes) != 0) {
_engineStatisticsPtr->SetLastError(
VE_SEND_ERROR, kTraceError,
"SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
return -1;
}
return 0;
}
int Channel::GetRemoteRTCPReportBlocks(
std::vector<ReportBlock>* report_blocks) {
if (report_blocks == NULL) {

41
voice_engine/channel.h
View File

@ -56,13 +56,37 @@ class RtpPacketReceived;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class TelephoneEventHandler;
class VoERTPObserver;
class VoiceEngineObserver;
struct CallStatistics;
struct ReportBlock;
struct SenderInfo;
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 6.4.2 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;
};
namespace voe {
class OutputMixer;
@ -215,10 +239,8 @@ class Channel
int SendTelephoneEventOutband(int event, int duration_ms);
int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
// VoERTP_RTCP
// RTP+RTCP
int SetLocalSSRC(unsigned int ssrc);
int GetLocalSSRC(unsigned int& ssrc);
int GetRemoteSSRC(unsigned int& ssrc);
int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
void EnableSendTransportSequenceNumber(int id);
@ -231,13 +253,7 @@ class Channel
void ResetSenderCongestionControlObjects();
void ResetReceiverCongestionControlObjects();
void SetRTCPStatus(bool enable);
int GetRTCPStatus(bool& enabled);
int SetRTCP_CNAME(const char cName[256]);
int GetRemoteRTCP_CNAME(char cName[256]);
int SendApplicationDefinedRTCPPacket(unsigned char subType,
unsigned int name,
const char* data,
unsigned short dataLengthInBytes);
int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
int GetRTPStatistics(CallStatistics& stats);
void SetNACKStatus(bool enable, int maxNumberOfPackets);
@ -343,6 +359,7 @@ class Channel
private:
class ProcessAndEncodeAudioTask;
int GetRemoteSSRC(unsigned int& ssrc);
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
bool OnRtpPacketWithHeader(const uint8_t* received_packet,

4
voice_engine/channel_proxy.h
View File

@ -18,8 +18,8 @@
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/channel.h"
#include "voice_engine/channel_manager.h"
#include "voice_engine/include/voe_rtp_rtcp.h"
#include <memory>
#include <string>
@ -42,8 +42,6 @@ class TransportFeedbackObserver;
namespace voe {
class Channel;
// This class provides the "view" of a voe::Channel that we need to implement
// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
// purposes:

151
voice_engine/include/voe_rtp_rtcp.h
View File

@ -1,151 +0,0 @@
/*
* 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 VOICE_ENGINE_VOE_RTP_RTCP_H_
#define VOICE_ENGINE_VOE_RTP_RTCP_H_
#include <vector>
#include "common_types.h" // NOLINT(build/include)
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 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;
protected:
VoERTP_RTCP() {}
virtual ~VoERTP_RTCP() {}
};
} // namespace webrtc
#endif // #ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_

201
voice_engine/voe_rtp_rtcp_impl.cc
View File

@ -1,201 +0,0 @@
/*
* 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.
*/
#include "system_wrappers/include/file_wrapper.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/include/voe_errors.h"
#include "voice_engine/voe_rtp_rtcp_impl.h"
#include "voice_engine/voice_engine_impl.h"
#include "voice_engine/channel.h"
#include "voice_engine/transmit_mixer.h"
namespace webrtc {
VoERTP_RTCP* VoERTP_RTCP::GetInterface(VoiceEngine* voiceEngine) {
if (NULL == voiceEngine) {
return NULL;
}
VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
s->AddRef();
return s;
}
VoERTP_RTCPImpl::VoERTP_RTCPImpl(voe::SharedData* shared) : _shared(shared) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
"VoERTP_RTCPImpl::VoERTP_RTCPImpl() - ctor");
}
VoERTP_RTCPImpl::~VoERTP_RTCPImpl() {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
"VoERTP_RTCPImpl::~VoERTP_RTCPImpl() - dtor");
}
int VoERTP_RTCPImpl::SetLocalSSRC(int channel, unsigned int ssrc) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetLocalSSRC(channel=%d, %lu)", channel, ssrc);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"SetLocalSSRC() failed to locate channel");
return -1;
}
return channelPtr->SetLocalSSRC(ssrc);
}
int VoERTP_RTCPImpl::GetLocalSSRC(int channel, unsigned int& ssrc) {
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"GetLocalSSRC() failed to locate channel");
return -1;
}
return channelPtr->GetLocalSSRC(ssrc);
}
int VoERTP_RTCPImpl::GetRemoteSSRC(int channel, unsigned int& ssrc) {
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"GetRemoteSSRC() failed to locate channel");
return -1;
}
return channelPtr->GetRemoteSSRC(ssrc);
}
int VoERTP_RTCPImpl::SetSendAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetSendAudioLevelIndicationStatus(channel=%d, enable=%d,"
" ID=%u)",
channel, enable, id);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
if (enable && (id < kVoiceEngineMinRtpExtensionId ||
id > kVoiceEngineMaxRtpExtensionId)) {
// [RFC5285] The 4-bit id is the local identifier of this element in
// the range 1-14 inclusive.
_shared->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"SetSendAudioLevelIndicationStatus() invalid ID parameter");
return -1;
}
// Set state and id for the specified channel.
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(
VE_CHANNEL_NOT_VALID, kTraceError,
"SetSendAudioLevelIndicationStatus() failed to locate channel");
return -1;
}
return channelPtr->SetSendAudioLevelIndicationStatus(enable, id);
}
int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"SetRTCPStatus() failed to locate channel");
return -1;
}
channelPtr->SetRTCPStatus(enable);
return 0;
}
int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled) {
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"GetRTCPStatus() failed to locate channel");
return -1;
}
return channelPtr->GetRTCPStatus(enabled);
}
int VoERTP_RTCPImpl::SetRTCP_CNAME(int channel, const char cName[256]) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"SetRTCP_CNAME(channel=%d, cName=%s)", channel, cName);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"SetRTCP_CNAME() failed to locate channel");
return -1;
}
return channelPtr->SetRTCP_CNAME(cName);
}
int VoERTP_RTCPImpl::GetRemoteRTCP_CNAME(int channel, char cName[256]) {
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"GetRemoteRTCP_CNAME() failed to locate channel");
return -1;
}
return channelPtr->GetRemoteRTCP_CNAME(cName);
}
int VoERTP_RTCPImpl::GetRTCPStatistics(int channel, CallStatistics& stats) {
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"GetRTPStatistics() failed to locate channel");
return -1;
}
return channelPtr->GetRTPStatistics(stats);
}
} // namespace webrtc

55
voice_engine/voe_rtp_rtcp_impl.h
View File

@ -1,55 +0,0 @@
/*
* 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.
*/
#ifndef VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H_
#define VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H_
#include "voice_engine/include/voe_rtp_rtcp.h"
#include "voice_engine/shared_data.h"
namespace webrtc {
class VoERTP_RTCPImpl : public VoERTP_RTCP {
public:
// RTCP
int SetRTCPStatus(int channel, bool enable) override;
int GetRTCPStatus(int channel, bool& enabled) override;
int SetRTCP_CNAME(int channel, const char cName[256]) override;
int GetRemoteRTCP_CNAME(int channel, char cName[256]) override;
// SSRC
int SetLocalSSRC(int channel, unsigned int ssrc) override;
int GetLocalSSRC(int channel, unsigned int& ssrc) override;
int GetRemoteSSRC(int channel, unsigned int& ssrc) override;
// RTP Header Extension for Client-to-Mixer Audio Level Indication
int SetSendAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id) override;
int GetRTCPStatistics(int channel, CallStatistics& stats) override;
protected:
VoERTP_RTCPImpl(voe::SharedData* shared);
~VoERTP_RTCPImpl() override;
private:
voe::SharedData* _shared;
};
} // namespace webrtc
#endif // VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H_

6
voice_engine/voice_engine_defines.h
View File

@ -56,12 +56,6 @@ enum { kVoiceEngineMinMinPlayoutDelayMs = 0 };
// Highest minimum playout delay
enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 };
// RTP/RTCP
// Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285)
enum { kVoiceEngineMinRtpExtensionId = 1 };
// Max 4-bit ID for RTP extension
enum { kVoiceEngineMaxRtpExtensionId = 14 };
} // namespace webrtc
namespace webrtc {

3
voice_engine/voice_engine_impl.h
View File

@ -17,7 +17,6 @@
#include "typedefs.h" // NOLINT(build/include)
#include "voice_engine/voe_base_impl.h"
#include "voice_engine/voe_network_impl.h"
#include "voice_engine/voe_rtp_rtcp_impl.h"
namespace webrtc {
namespace voe {
@ -27,13 +26,11 @@ class ChannelProxy;
class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
public VoiceEngine,
public VoENetworkImpl,
public VoERTP_RTCPImpl,
public VoEBaseImpl {
public:
VoiceEngineImpl()
: SharedData(),
VoENetworkImpl(this),
VoERTP_RTCPImpl(this),
VoEBaseImpl(this),
_ref_count(0) {}
~VoiceEngineImpl() override { assert(_ref_count.Value() == 0); }