Remove VoERTP_RTCP.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/3006383002
Cr-Commit-Position: refs/heads/master@{#19892}
diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h
index e388993..45a9f9f 100644
--- a/test/mock_voice_engine.h
+++ b/test/mock_voice_engine.h
@@ -127,44 +127,6 @@
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
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 5f1ae03..600645e 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -19,7 +19,6 @@
"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 @@
"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",
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 49bc381..a44ccb8 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -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 @@
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 @@
_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 @@
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) {
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 5c95a64..48aeafe 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -56,13 +56,37 @@
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 @@
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 @@
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 @@
private:
class ProcessAndEncodeAudioTask;
+ int GetRemoteSSRC(unsigned int& ssrc);
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
bool OnRtpPacketWithHeader(const uint8_t* received_packet,
diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h
index ca11b58..40246aaa 100644
--- a/voice_engine/channel_proxy.h
+++ b/voice_engine/channel_proxy.h
@@ -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 @@
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:
diff --git a/voice_engine/include/voe_rtp_rtcp.h b/voice_engine/include/voe_rtp_rtcp.h
deleted file mode 100644
index e863a28..0000000
--- a/voice_engine/include/voe_rtp_rtcp.h
+++ /dev/null
@@ -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_
diff --git a/voice_engine/voe_rtp_rtcp_impl.cc b/voice_engine/voe_rtp_rtcp_impl.cc
deleted file mode 100644
index 15d5022..0000000
--- a/voice_engine/voe_rtp_rtcp_impl.cc
+++ /dev/null
@@ -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
diff --git a/voice_engine/voe_rtp_rtcp_impl.h b/voice_engine/voe_rtp_rtcp_impl.h
deleted file mode 100644
index b640bdf..0000000
--- a/voice_engine/voe_rtp_rtcp_impl.h
+++ /dev/null
@@ -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_
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index c16dd56..bc5eb1b 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -56,12 +56,6 @@
// 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 {
diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h
index ef03e70..2ede6c3 100644
--- a/voice_engine/voice_engine_impl.h
+++ b/voice_engine/voice_engine_impl.h
@@ -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 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); }
