Remove voe::Statistics.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3020473002
Cr-Commit-Position: refs/heads/master@{#20042}

audio/audio_receive_stream.cc

@@ -94,7 +94,7 @@ AudioReceiveStream::AudioReceiveStream(
   RTC_CHECK_EQ(config.decoder_factory,
                channel_proxy_->GetAudioDecoderFactory());
 
-  channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
+  channel_proxy_->RegisterTransport(config.rtcp_send_transport);
   channel_proxy_->SetReceiveCodecs(config.decoder_map);
 
   for (const auto& extension : config.rtp.extensions) {
@@ -122,7 +122,7 @@ AudioReceiveStream::~AudioReceiveStream() {
     Stop();
   }
   channel_proxy_->DisassociateSendChannel();
-  channel_proxy_->DeRegisterExternalTransport();
+  channel_proxy_->RegisterTransport(nullptr);
   channel_proxy_->ResetReceiverCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
 }

audio/audio_receive_stream_unittest.cc

@@ -101,10 +101,7 @@ struct ConfigHelper {
                   .Times(1);
           EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
               .Times(1);
-          EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
+          EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2);
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
-              .Times(1);
           EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
               .WillOnce(ReturnRef(decoder_factory_));
           testing::Expectation expect_set =

audio/audio_send_stream.cc

@@ -120,7 +120,7 @@ AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
-  channel_proxy_->DeRegisterExternalTransport();
+  channel_proxy_->RegisterTransport(nullptr);
   channel_proxy_->ResetSenderCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
   channel_proxy_->SetRtcpRttStats(nullptr);
@@ -164,7 +164,7 @@ void AudioSendStream::ConfigureStream(
   if (first_time ||
       new_config.send_transport != old_config.send_transport) {
     if (old_config.send_transport) {
-      channel_proxy->DeRegisterExternalTransport();
+      channel_proxy->RegisterTransport(nullptr);
     }
     if (new_config.send_transport) {
       stream->timed_send_transport_adapter_.reset(new TimedTransport(
@@ -172,7 +172,7 @@ void AudioSendStream::ConfigureStream(
     } else {
       stream->timed_send_transport_adapter_.reset(nullptr);
     }
-    channel_proxy->RegisterExternalTransport(
+    channel_proxy->RegisterTransport(
         stream->timed_send_transport_adapter_.get());
   }
 

audio/audio_send_stream_unittest.cc

@@ -235,8 +235,7 @@ struct ConfigHelper {
         .Times(1);
     EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
         .Times(1);
-    EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);
+    EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2);
-    EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
         .Times(1);  // Destructor resets the event log

test/mock_voe_channel_proxy.h

@@ -65,8 +65,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   // TODO(solenberg): Talk the compiler into accepting this mock method:
   // MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink));
   MOCK_METHOD1(SetInputMute, void(bool muted));
-  MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
+  MOCK_METHOD1(RegisterTransport, void(Transport* transport));
-  MOCK_METHOD0(DeRegisterExternalTransport, void());
   MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
   MOCK_CONST_METHOD0(GetAudioDecoderFactory,

voice_engine/BUILD.gn

@@ -20,8 +20,6 @@ rtc_static_library("voice_engine") {
     "include/voe_errors.h",
     "shared_data.cc",
     "shared_data.h",
-    "statistics.cc",
-    "statistics.h",
     "transmit_mixer.cc",
     "transmit_mixer.h",
     "transport_feedback_packet_loss_tracker.cc",

voice_engine/channel.cc

@@ -40,7 +40,6 @@
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 #include "system_wrappers/include/trace.h"
-#include "voice_engine/statistics.h"
 #include "voice_engine/utility.h"
 
 namespace webrtc {
@@ -447,9 +446,8 @@ int32_t Channel::SendData(FrameType frameType,
           // received from the capture device as
           // undefined for voice for now.
           -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) <<
-        VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
+        "Channel::SendData() failed to send data to RTP/RTCP module";
-        "Channel::SendData() failed to send data to RTP/RTCP module");
     return -1;
   }
 
@@ -475,11 +473,7 @@ bool Channel::SendRtp(const uint8_t* data,
   size_t bufferLength = len;
 
   if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
-    std::string transport_name =
+    LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
-        _externalTransport ? "external transport" : "WebRtc sockets";
-    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::SendPacket() RTP transmission using %s failed",
-                 transport_name.c_str());
     return false;
   }
   return true;
@@ -502,11 +496,7 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) {
 
   int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
   if (n < 0) {
-    std::string transport_name =
+    LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
-        _externalTransport ? "external transport" : "WebRtc sockets";
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::SendRtcp() transmission using %s failed",
-                 transport_name.c_str());
     return false;
   }
   return true;
@@ -556,7 +546,6 @@ int32_t Channel::OnInitializeDecoder(
                  "Channel::OnInitializeDecoder() invalid codec ("
                  "pt=%d, name=%s) received - 1",
                  payloadType, payloadName);
-    _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
     return -1;
   }
 
@@ -585,9 +574,8 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
   // Push the incoming payload (parsed and ready for decoding) into the ACM
   if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) !=
       0) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) <<
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
+        "Channel::OnReceivedPayloadData() unable to push data to the ACM";
-        "Channel::OnReceivedPayloadData() unable to push data to the ACM");
     return -1;
   }
 
@@ -759,7 +747,6 @@ Channel::Channel(int32_t channelId,
                                            rtp_payload_registry_.get())),
       telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
       _outputAudioLevel(),
-      _externalTransport(false),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
       ntp_estimator_(Clock::GetRealTimeClock()),
@@ -769,10 +756,8 @@ Channel::Channel(int32_t channelId,
       rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
       capture_start_rtp_time_stamp_(-1),
       capture_start_ntp_time_ms_(-1),
-      _engineStatisticsPtr(NULL),
       _moduleProcessThreadPtr(NULL),
       _audioDeviceModulePtr(NULL),
-      _callbackCritSectPtr(NULL),
       _transportPtr(NULL),
       input_mute_(false),
       previous_frame_muted_(false),
@@ -835,7 +820,7 @@ int32_t Channel::Init() {
 
   // --- Initial sanity
 
-  if ((_engineStatisticsPtr == NULL) || (_moduleProcessThreadPtr == NULL)) {
+  if (_moduleProcessThreadPtr == NULL) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::Init() must call SetEngineInformation() first");
     return -1;
@@ -848,9 +833,7 @@ int32_t Channel::Init() {
   // --- ACM initialization
 
   if (audio_coding_->InitializeReceiver() == -1) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "Channel::Init() unable to initialize the ACM - 1";
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "Channel::Init() unable to initialize the ACM - 1");
     return -1;
   }
 
@@ -866,9 +849,7 @@ int32_t Channel::Init() {
   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
   // --- Register all permanent callbacks
   if (audio_coding_->RegisterTransportCallback(this) == -1) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "Channel::Init() callbacks not registered";
-        VE_CANNOT_INIT_CHANNEL, kTraceError,
-        "Channel::Init() callbacks not registered");
     return -1;
   }
 
@@ -903,19 +884,15 @@ void Channel::Terminate() {
   // End of modules shutdown
 }
 
-int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
+int32_t Channel::SetEngineInformation(ProcessThread& moduleProcessThread,
-                                      ProcessThread& moduleProcessThread,
                                       AudioDeviceModule& audioDeviceModule,
-                                      rtc::CriticalSection* callbackCritSect,
                                       rtc::TaskQueue* encoder_queue) {
   RTC_DCHECK(encoder_queue);
   RTC_DCHECK(!encoder_queue_);
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetEngineInformation()");
-  _engineStatisticsPtr = &engineStatistics;
   _moduleProcessThreadPtr = &moduleProcessThread;
   _audioDeviceModulePtr = &audioDeviceModule;
-  _callbackCritSectPtr = callbackCritSect;
   encoder_queue_ = encoder_queue;
   return 0;
 }
@@ -974,9 +951,7 @@ int32_t Channel::StartSend() {
   }
   _rtpRtcpModule->SetSendingMediaStatus(true);
   if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
-        VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-        "StartSend() RTP/RTCP failed to start sending");
     _rtpRtcpModule->SetSendingMediaStatus(false);
     rtc::CritScope cs(&_callbackCritSect);
     channel_state_.SetSending(false);
@@ -1023,9 +998,7 @@ void Channel::StopSend() {
   // Reset sending SSRC and sequence number and triggers direct transmission
   // of RTCP BYE
   if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
-        VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
-        "StartSend() RTP/RTCP failed to stop sending");
   }
   _rtpRtcpModule->SetSendingMediaStatus(false);
 }
@@ -1198,39 +1171,9 @@ void Channel::SetReceiverFrameLengthRange(int min_frame_length_ms,
   });
 }
 
-int32_t Channel::RegisterExternalTransport(Transport* transport) {
+void Channel::RegisterTransport(Transport* transport) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::RegisterExternalTransport()");
-
   rtc::CritScope cs(&_callbackCritSect);
-  if (_externalTransport) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_OPERATION, kTraceError,
-        "RegisterExternalTransport() external transport already enabled");
-    return -1;
-  }
-  _externalTransport = true;
   _transportPtr = transport;
-  return 0;
-}
-
-int32_t Channel::DeRegisterExternalTransport() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::DeRegisterExternalTransport()");
-
-  rtc::CritScope cs(&_callbackCritSect);
-  if (_transportPtr) {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "DeRegisterExternalTransport() all transport is disabled");
-  } else {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_OPERATION, kTraceWarning,
-        "DeRegisterExternalTransport() external transport already "
-        "disabled");
-  }
-  _externalTransport = false;
-  _transportPtr = NULL;
-  return 0;
 }
 
 void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
@@ -1380,9 +1323,7 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
   }
   if (_rtpRtcpModule->SendTelephoneEventOutband(
       event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
-        VE_SEND_DTMF_FAILED, kTraceWarning,
-        "SendTelephoneEventOutband() failed to send event");
     return -1;
   }
   return 0;
@@ -1401,10 +1342,8 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type,
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
     if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
-      _engineStatisticsPtr->SetLastError(
+      LOG(LS_ERROR) << "SetSendTelephoneEventPayloadType() failed to register "
-          VE_RTP_RTCP_MODULE_ERROR, kTraceError,
+                       "send payload type";
-          "SetSendTelephoneEventPayloadType() failed to register send"
-          "payload type");
       return -1;
     }
   }
@@ -1415,8 +1354,7 @@ int Channel::SetLocalSSRC(unsigned int ssrc) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetLocalSSRC()");
   if (channel_state_.Get().sending) {
-    _engineStatisticsPtr->SetLastError(VE_ALREADY_SENDING, kTraceError,
+    LOG(LS_ERROR) << "SetLocalSSRC() already sending";
-                                       "SetLocalSSRC() already sending");
     return -1;
   }
   _rtpRtcpModule->SetSSRC(ssrc);
@@ -1517,9 +1455,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRTCP_CNAME()");
   if (_rtpRtcpModule->SetCNAME(cName) != 0) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
-        VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-        "SetRTCP_CNAME() failed to set RTCP CNAME");
     return -1;
   }
   return 0;
@@ -1528,9 +1464,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) {
 int Channel::GetRemoteRTCPReportBlocks(
     std::vector<ReportBlock>* report_blocks) {
   if (report_blocks == NULL) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
-        VE_INVALID_ARGUMENT, kTraceError,
-        "GetRemoteRTCPReportBlock()s invalid report_blocks.");
     return -1;
   }
 
@@ -1788,15 +1722,11 @@ int Channel::SetMinimumPlayoutDelay(int delayMs) {
                "Channel::SetMinimumPlayoutDelay()");
   if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
       (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetMinimumPlayoutDelay() invalid min delay");
     return -1;
   }
   if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "SetMinimumPlayoutDelay() failed to set min playout delay";
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "SetMinimumPlayoutDelay() failed to set min playout delay");
     return -1;
   }
   return 0;
@@ -1809,9 +1739,7 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
     playout_timestamp_rtp = playout_timestamp_rtp_;
   }
   if (playout_timestamp_rtp == 0) {
-    _engineStatisticsPtr->SetLastError(
+    LOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp";
-        VE_CANNOT_RETRIEVE_VALUE, kTraceStateInfo,
-        "GetPlayoutTimestamp() failed to retrieve timestamp");
     return -1;
   }
   timestamp = playout_timestamp_rtp;
@@ -1839,9 +1767,6 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::UpdatePlayoutTimestamp() failed to read playout"
                  " delay from the ADM");
-    _engineStatisticsPtr->SetLastError(
-        VE_CANNOT_RETRIEVE_VALUE, kTraceError,
-        "UpdatePlayoutTimestamp() failed to retrieve playout delay");
     return;
   }
 

voice_engine/channel.h

@@ -90,7 +90,6 @@ namespace voe {
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
-class Statistics;
 class TransportFeedbackProxy;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
@@ -157,10 +156,8 @@ class Channel
           const VoEBase::ChannelConfig& config);
   int32_t Init();
   void Terminate();
-  int32_t SetEngineInformation(Statistics& engineStatistics,
+  int32_t SetEngineInformation(ProcessThread& moduleProcessThread,
-                               ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
-                               rtc::CriticalSection* callbackCritSect,
                                rtc::TaskQueue* encoder_queue);
 
   void SetSink(std::unique_ptr<AudioSinkInterface> sink);
@@ -197,8 +194,7 @@ class Channel
                                    int max_frame_length_ms);
 
   // Network
-  int32_t RegisterExternalTransport(Transport* transport);
+  void RegisterTransport(Transport* transport);
-  int32_t DeRegisterExternalTransport();
   // TODO(nisse, solenberg): Delete when VoENetwork is deleted.
   int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
   void OnRtpPacket(const RtpPacketReceived& packet);
@@ -286,10 +282,6 @@ class Channel
   int32_t ChannelId() const { return _channelId; }
   bool Playing() const { return channel_state_.Get().playing; }
   bool Sending() const { return channel_state_.Get().sending; }
-  bool ExternalTransport() const {
-    rtc::CritScope cs(&_callbackCritSect);
-    return _externalTransport;
-  }
   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
   int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
 
@@ -395,7 +387,6 @@ class Channel
   acm2::RentACodec rent_a_codec_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
-  bool _externalTransport;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_);
@@ -420,10 +411,8 @@ class Channel
   int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
 
   // uses
-  Statistics* _engineStatisticsPtr;
   ProcessThread* _moduleProcessThreadPtr;
   AudioDeviceModule* _audioDeviceModulePtr;
-  rtc::CriticalSection* _callbackCritSectPtr;    // owned by base
   Transport* _transportPtr;  // WebRtc socket or external transport
   RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_);
   bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);

voice_engine/channel_proxy.cc

@@ -213,15 +213,9 @@ void ChannelProxy::SetInputMute(bool muted) {
   channel()->SetInputMute(muted);
 }
 
-void ChannelProxy::RegisterExternalTransport(Transport* transport) {
+void ChannelProxy::RegisterTransport(Transport* transport) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  int error = channel()->RegisterExternalTransport(transport);
+  channel()->RegisterTransport(transport);
-  RTC_DCHECK_EQ(0, error);
-}
-
-void ChannelProxy::DeRegisterExternalTransport() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  channel()->DeRegisterExternalTransport();
 }
 
 void ChannelProxy::OnRtpPacket(const RtpPacketReceived& packet) {

voice_engine/channel_proxy.h

@@ -94,8 +94,7 @@ class ChannelProxy : public RtpPacketSinkInterface {
   virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
-  virtual void RegisterExternalTransport(Transport* transport);
+  virtual void RegisterTransport(Transport* transport);
-  virtual void DeRegisterExternalTransport();
 
   // Implements RtpPacketSinkInterface
   void OnRtpPacket(const RtpPacketReceived& packet) override;

voice_engine/shared_data.cc

@@ -24,7 +24,6 @@ static int32_t _gInstanceCounter = 0;
 SharedData::SharedData()
     : _instanceId(++_gInstanceCounter),
       _channelManager(_gInstanceCounter),
-      _engineStatistics(_gInstanceCounter),
       _audioDevicePtr(NULL),
       _moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")),
       encoder_queue_("AudioEncoderQueue") {
@@ -84,20 +83,6 @@ int SharedData::NumOfPlayingChannels() {
   return playout_channels;
 }
 
-void SharedData::SetLastError(int32_t error) const {
-  _engineStatistics.SetLastError(error);
-}
-
-void SharedData::SetLastError(int32_t error,
-                              TraceLevel level) const {
-  _engineStatistics.SetLastError(error, level);
-}
-
-void SharedData::SetLastError(int32_t error, TraceLevel level,
-                              const char* msg) const {
-  _engineStatistics.SetLastError(error, level, msg);
-}
-
 }  // namespace voe
 
 }  // namespace webrtc

voice_engine/shared_data.h

@@ -22,7 +22,6 @@
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/thread_checker.h"
 #include "voice_engine/channel_manager.h"
-#include "voice_engine/statistics.h"
 #include "voice_engine/voice_engine_defines.h"
 
 class ProcessThread;
@@ -37,7 +36,6 @@ class SharedData
 public:
     // Public accessors.
     uint32_t instance_id() const { return _instanceId; }
-    Statistics& statistics() { return _engineStatistics; }
     ChannelManager& channel_manager() { return _channelManager; }
     AudioDeviceModule* audio_device() { return _audioDevicePtr.get(); }
     void set_audio_device(
@@ -51,18 +49,11 @@ public:
     int NumOfSendingChannels();
     int NumOfPlayingChannels();
 
-    // Convenience methods for calling statistics().SetLastError().
-    void SetLastError(int32_t error) const;
-    void SetLastError(int32_t error, TraceLevel level) const;
-    void SetLastError(int32_t error, TraceLevel level,
-                      const char* msg) const;
-
 protected:
  rtc::ThreadChecker construction_thread_;
  const uint32_t _instanceId;
  rtc::CriticalSection _apiCritPtr;
  ChannelManager _channelManager;
- Statistics _engineStatistics;
  rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
  TransmitMixer* _transmitMixerPtr;
  std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;

voice_engine/statistics.cc

@@ -1,86 +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 <assert.h>
-#include <stdio.h>
-
-#include "voice_engine/statistics.h"
-
-#include "system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-namespace voe {
-
-Statistics::Statistics(uint32_t instanceId) :
-    _instanceId(instanceId),
-    _lastError(0),
-    _isInitialized(false)
-{
-}
-
-Statistics::~Statistics()
-{
-}
-
-int32_t Statistics::SetInitialized()
-{
-    _isInitialized = true;
-    return 0;
-}
-
-int32_t Statistics::SetUnInitialized()
-{
-    _isInitialized = false;
-    return 0;
-}
-
-bool Statistics::Initialized() const
-{
-    return _isInitialized;
-}
-
-int32_t Statistics::SetLastError(int32_t error) const
-{
-    rtc::CritScope cs(&lock_);
-    _lastError = error;
-    return 0;
-}
-
-int32_t Statistics::SetLastError(int32_t error,
-                                 TraceLevel level) const
-{
-    WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1),
-                 "error code is set to %d",
-                 error);
-    rtc::CritScope cs(&lock_);
-    _lastError = error;
-    return 0;
-}
-
-int32_t Statistics::SetLastError(
-    int32_t error,
-    TraceLevel level, const char* msg) const
-{
-    char traceMessage[KTraceMaxMessageSize];
-    assert(strlen(msg) < KTraceMaxMessageSize);
-    sprintf(traceMessage, "%s (error=%d)", msg, error);
-
-    WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1), "%s",
-                 traceMessage);
-
-    rtc::CritScope cs(&lock_);
-    _lastError = error;
-    return 0;
-}
-
-}  // namespace voe
-
-}  // namespace webrtc

voice_engine/statistics.h

@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2011 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_STATISTICS_H_
-#define VOICE_ENGINE_STATISTICS_H_
-
-#include "common_types.h"  // NOLINT(build/include)
-#include "rtc_base/criticalsection.h"
-#include "typedefs.h"  // NOLINT(build/include)
-#include "voice_engine/include/voe_errors.h"
-#include "voice_engine/voice_engine_defines.h"
-
-namespace webrtc {
-namespace voe {
-
-class Statistics
-{
- public:
-    enum {KTraceMaxMessageSize = 256};
- public:
-    Statistics(uint32_t instanceId);
-    ~Statistics();
-
-    int32_t SetInitialized();
-    int32_t SetUnInitialized();
-    bool Initialized() const;
-    int32_t SetLastError(int32_t error) const;
-    int32_t SetLastError(int32_t error, TraceLevel level) const;
-    int32_t SetLastError(int32_t error,
-                         TraceLevel level,
-                         const char* msg) const;
-
- private:
-    rtc::CriticalSection lock_;
-    const uint32_t _instanceId;
-    mutable int32_t _lastError;
-    bool _isInitialized;
-};
-
-}  // namespace voe
-
-}  // namespace webrtc
-
-#endif // VOICE_ENGINE_STATISTICS_H_

voice_engine/transmit_mixer.cc

@@ -20,7 +20,6 @@
 #include "system_wrappers/include/trace.h"
 #include "voice_engine/channel.h"
 #include "voice_engine/channel_manager.h"
-#include "voice_engine/statistics.h"
 #include "voice_engine/utility.h"
 
 namespace webrtc {

voice_engine/transmit_mixer.h

@@ -36,7 +36,6 @@ namespace voe {
 
 class ChannelManager;
 class MixedAudio;
-class Statistics;
 
 class TransmitMixer {
 public:

voice_engine/voe_base_impl.cc

@@ -42,7 +42,6 @@ VoEBaseImpl::~VoEBaseImpl() {
 }
 
 void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
-  rtc::CritScope cs(&callbackCritSect_);
   if (error == AudioDeviceObserver::kRecordingError) {
     LOG_F(LS_ERROR) << "VE_RUNTIME_REC_ERROR";
   } else if (error == AudioDeviceObserver::kPlayoutError) {
@@ -51,7 +50,6 @@ void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
 }
 
 void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) {
-  rtc::CritScope cs(&callbackCritSect_);
   if (warning == AudioDeviceObserver::kRecordingWarning) {
     LOG_F(LS_WARNING) << "VE_RUNTIME_REC_WARNING";
   } else if (warning == AudioDeviceObserver::kPlayoutWarning) {
@@ -167,9 +165,6 @@ int VoEBaseImpl::Init(
   RTC_DCHECK(audio_processing);
   rtc::CritScope cs(shared_->crit_sec());
   WebRtcSpl_Init();
-  if (shared_->statistics().Initialized()) {
-    return 0;
-  }
   if (shared_->process_thread()) {
     shared_->process_thread()->Start();
   }
@@ -185,8 +180,7 @@ int VoEBaseImpl::Init(
         VoEId(shared_->instance_id(), -1),
         AudioDeviceModule::kPlatformDefaultAudio));
     if (shared_->audio_device() == nullptr) {
-      shared_->SetLastError(VE_NO_MEMORY, kTraceCritical,
+      LOG(LS_ERROR) << "Init() failed to create the ADM";
-                            "Init() failed to create the ADM");
       return -1;
     }
 #endif  // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
@@ -211,55 +205,44 @@ int VoEBaseImpl::Init(
 
   // Register the AudioObserver implementation
   if (shared_->audio_device()->RegisterEventObserver(this) != 0) {
-    shared_->SetLastError(
+    LOG(LS_ERROR) << "Init() failed to register event observer for the ADM";
-        VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-        "Init() failed to register event observer for the ADM");
   }
 
   // Register the AudioTransport implementation
   if (shared_->audio_device()->RegisterAudioCallback(this) != 0) {
-    shared_->SetLastError(
+    LOG(LS_ERROR) << "Init() failed to register audio callback for the ADM";
-        VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-        "Init() failed to register audio callback for the ADM");
   }
 
   // ADM initialization
   if (shared_->audio_device()->Init() != 0) {
-    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+    LOG(LS_ERROR) << "Init() failed to initialize the ADM";
-                          "Init() failed to initialize the ADM");
     return -1;
   }
 
   // Initialize the default speaker
   if (shared_->audio_device()->SetPlayoutDevice(
           WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
-    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
+    LOG(LS_ERROR) << "Init() failed to set the default output device";
-                          "Init() failed to set the default output device");
   }
   if (shared_->audio_device()->InitSpeaker() != 0) {
-    shared_->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
+    LOG(LS_ERROR) << "Init() failed to initialize the speaker";
-                          "Init() failed to initialize the speaker");
   }
 
   // Initialize the default microphone
   if (shared_->audio_device()->SetRecordingDevice(
           WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
-    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
+    LOG(LS_ERROR) << "Init() failed to set the default input device";
-                          "Init() failed to set the default input device");
   }
   if (shared_->audio_device()->InitMicrophone() != 0) {
-    shared_->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
+    LOG(LS_ERROR) << "Init() failed to initialize the microphone";
-                          "Init() failed to initialize the microphone");
   }
 
   // Set number of channels
   if (shared_->audio_device()->StereoPlayoutIsAvailable(&available) != 0) {
-    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+    LOG(LS_ERROR) << "Init() failed to query stereo playout mode";
-                          "Init() failed to query stereo playout mode");
   }
   if (shared_->audio_device()->SetStereoPlayout(available) != 0) {
-    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+    LOG(LS_ERROR) << "Init() failed to set mono/stereo playout mode";
-                          "Init() failed to set mono/stereo playout mode");
   }
 
   // TODO(andrew): These functions don't tell us whether stereo recording
@@ -271,14 +254,11 @@ int VoEBaseImpl::Init(
   // http://code.google.com/p/webrtc/issues/detail?id=204
   shared_->audio_device()->StereoRecordingIsAvailable(&available);
   if (shared_->audio_device()->SetStereoRecording(available) != 0) {
-    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+    LOG(LS_ERROR) << "Init() failed to set mono/stereo recording mode";
-                          "Init() failed to set mono/stereo recording mode");
   }
 
   shared_->set_audio_processing(audio_processing);
 
-  // Set the error state for any failures in this block.
-  shared_->SetLastError(VE_APM_ERROR);
   // Configure AudioProcessing components.
   // TODO(peah): Move this initialization to webrtcvoiceengine.cc.
   if (audio_processing->high_pass_filter()->Enable(true) != 0) {
@@ -309,14 +289,12 @@ int VoEBaseImpl::Init(
     LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState;
     return -1;
   }
-  shared_->SetLastError(0);  // Clear error state.
 
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   bool agc_enabled =
       agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled();
   if (shared_->audio_device()->SetAGC(agc_enabled) != 0) {
     LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled;
-    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR);
     // TODO(ajm): No error return here due to
     // https://code.google.com/p/webrtc/issues/detail?id=1464
   }
@@ -327,7 +305,7 @@ int VoEBaseImpl::Init(
   else
     decoder_factory_ = CreateBuiltinAudioDecoderFactory();
 
-  return shared_->statistics().SetInitialized();
+  return 0;
 }
 
 int VoEBaseImpl::Terminate() {
@@ -341,11 +319,6 @@ int VoEBaseImpl::CreateChannel() {
 
 int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
   rtc::CritScope cs(shared_->crit_sec());
-  if (!shared_->statistics().Initialized()) {
-    shared_->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-
   ChannelConfig config_copy(config);
   config_copy.acm_config.decoder_factory = decoder_factory_;
   voe::ChannelOwner channel_owner =
@@ -355,21 +328,16 @@ int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
 
 int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
   if (channel_owner->channel()->SetEngineInformation(
-          shared_->statistics(),
           *shared_->process_thread(), *shared_->audio_device(),
-          &callbackCritSect_, shared_->encoder_queue()) != 0) {
+          shared_->encoder_queue()) != 0) {
-    shared_->SetLastError(
+    LOG(LS_ERROR) << "CreateChannel() failed to associate engine and channel."
-        VE_CHANNEL_NOT_CREATED, kTraceError,
+                     " Destroying channel.";
-        "CreateChannel() failed to associate engine and channel."
-        " Destroying channel.");
     shared_->channel_manager().DestroyChannel(
         channel_owner->channel()->ChannelId());
     return -1;
   } else if (channel_owner->channel()->Init() != 0) {
-    shared_->SetLastError(
+    LOG(LS_ERROR) << "CreateChannel() failed to initialize channel. Destroying"
-        VE_CHANNEL_NOT_CREATED, kTraceError,
+                     " channel.";
-        "CreateChannel() failed to initialize channel. Destroying"
-        " channel.");
     shared_->channel_manager().DestroyChannel(
         channel_owner->channel()->ChannelId());
     return -1;
@@ -379,17 +347,11 @@ int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
 
 int VoEBaseImpl::DeleteChannel(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
-  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 == nullptr) {
-      shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+      LOG(LS_ERROR) << "DeleteChannel() failed to locate channel";
-                            "DeleteChannel() failed to locate channel");
       return -1;
     }
   }
@@ -406,23 +368,17 @@ int VoEBaseImpl::DeleteChannel(int channel) {
 
 int VoEBaseImpl::StartPlayout(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
-  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 == nullptr) {
-    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    LOG(LS_ERROR) << "StartPlayout() failed to locate channel";
-                          "StartPlayout() failed to locate channel");
     return -1;
   }
   if (channelPtr->Playing()) {
     return 0;
   }
   if (StartPlayout() != 0) {
-    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+    LOG(LS_ERROR) << "StartPlayout() failed to start playout";
-                          "StartPlayout() failed to start playout");
     return -1;
   }
   return channelPtr->StartPlayout();
@@ -430,15 +386,10 @@ int VoEBaseImpl::StartPlayout(int channel) {
 
 int VoEBaseImpl::StopPlayout(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
-  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 == nullptr) {
-    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    LOG(LS_ERROR) << "StopPlayout() failed to locate channel";
-                          "StopPlayout() failed to locate channel");
     return -1;
   }
   if (channelPtr->StopPlayout() != 0) {
@@ -450,23 +401,17 @@ int VoEBaseImpl::StopPlayout(int channel) {
 
 int VoEBaseImpl::StartSend(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
-  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 == nullptr) {
-    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    LOG(LS_ERROR) << "StartSend() failed to locate channel";
-                          "StartSend() failed to locate channel");
     return -1;
   }
   if (channelPtr->Sending()) {
     return 0;
   }
   if (StartSend() != 0) {
-    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+    LOG(LS_ERROR) << "StartSend() failed to start recording";
-                          "StartSend() failed to start recording");
     return -1;
   }
   return channelPtr->StartSend();
@@ -474,15 +419,10 @@ int VoEBaseImpl::StartSend(int channel) {
 
 int VoEBaseImpl::StopSend(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
-  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 == nullptr) {
-    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    LOG(LS_ERROR) << "StopSend() failed to locate channel";
-                          "StopSend() failed to locate channel");
     return -1;
   }
   channelPtr->StopSend();
@@ -507,8 +447,7 @@ int32_t VoEBaseImpl::StopPlayout() {
   // Stop audio-device playing if no channel is playing out
   if (shared_->NumOfPlayingChannels() == 0) {
     if (shared_->audio_device()->StopPlayout() != 0) {
-      shared_->SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError,
+      LOG(LS_ERROR) << "StopPlayout() failed to stop playout";
-                            "StopPlayout() failed to stop playout");
       return -1;
     }
   }
@@ -536,8 +475,7 @@ int32_t VoEBaseImpl::StopSend() {
   if (shared_->NumOfSendingChannels() == 0) {
     // Stop audio-device recording if no channel is recording
     if (shared_->audio_device()->StopRecording() != 0) {
-      shared_->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError,
+      LOG(LS_ERROR) << "StopSend() failed to stop recording";
-                            "StopSend() failed to stop recording");
       return -1;
     }
     shared_->transmit_mixer()->StopSend();
@@ -559,34 +497,27 @@ int32_t VoEBaseImpl::TerminateInternal() {
 
   if (shared_->audio_device()) {
     if (shared_->audio_device()->StopPlayout() != 0) {
-      shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+      LOG(LS_ERROR) << "TerminateInternal() failed to stop playout";
-                            "TerminateInternal() failed to stop playout");
     }
     if (shared_->audio_device()->StopRecording() != 0) {
-      shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+      LOG(LS_ERROR) << "TerminateInternal() failed to stop recording";
-                            "TerminateInternal() failed to stop recording");
     }
     if (shared_->audio_device()->RegisterEventObserver(nullptr) != 0) {
-      shared_->SetLastError(
+      LOG(LS_ERROR) << "TerminateInternal() failed to de-register event "
-          VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+                       "observer for the ADM";
-          "TerminateInternal() failed to de-register event observer "
-          "for the ADM");
     }
     if (shared_->audio_device()->RegisterAudioCallback(nullptr) != 0) {
-      shared_->SetLastError(
+      LOG(LS_ERROR) << "TerminateInternal() failed to de-register audio "
-          VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+                       "callback for the ADM";
-          "TerminateInternal() failed to de-register audio callback "
-          "for the ADM");
     }
     if (shared_->audio_device()->Terminate() != 0) {
-      shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+      LOG(LS_ERROR) << "TerminateInternal() failed to terminate the ADM";
-                            "TerminateInternal() failed to terminate the ADM");
     }
     shared_->set_audio_device(nullptr);
   }
 
   shared_->set_audio_processing(nullptr);
 
-  return shared_->statistics().SetUnInitialized();
+  return 0;
 }
 }  // namespace webrtc

voice_engine/voe_base_impl.h

@@ -104,7 +104,6 @@ class VoEBaseImpl : public VoEBase,
   // Initialize channel by setting Engine Information then initializing
   // channel.
   int InitializeChannel(voe::ChannelOwner* channel_owner);
-  rtc::CriticalSection callbackCritSect_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 
   AudioFrame audioFrame_;

voice_engine/voe_base_unittest.cc

@@ -41,11 +41,6 @@ TEST_F(VoEBaseTest, InitWithExternalAudioDevice) {
   EXPECT_EQ(0, base_->Init(&adm_, apm_.get()));
 }
 
-TEST_F(VoEBaseTest, CreateChannelBeforeInitShouldFail) {
-  int channelID = base_->CreateChannel();
-  EXPECT_EQ(channelID, -1);
-}
-
 TEST_F(VoEBaseTest, CreateChannelAfterInit) {
   EXPECT_EQ(0, base_->Init(&adm_, apm_.get(), nullptr));
   int channelID = base_->CreateChannel();

voice_engine/voice_engine_impl.cc

@@ -65,7 +65,6 @@ std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
     int channel_id) {
   RTC_DCHECK(channel_id >= 0);
   rtc::CritScope cs(crit_sec());
-  RTC_DCHECK(statistics().Initialized());
   return std::unique_ptr<voe::ChannelProxy>(
       new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
 }