Delete AudioMonitor and related code.

Bug: webrtc:8760
Change-Id: I0b11ec66b0f2576f52866864ba046191034a4d2d
Reviewed-on: https://webrtc-review.googlesource.com/39003
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21801}

audio/audio_level.cc

@@ -16,25 +16,13 @@
 namespace webrtc {
 namespace voe {
 
-// Number of bars on the indicator.
-// Note that the number of elements is specified because we are indexing it
-// in the range of 0-32
-constexpr int8_t kPermutation[33] = {0, 1, 2, 3, 4, 4, 5, 5, 5, 5, 6,
-                                     6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8,
-                                     9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9};
-
 AudioLevel::AudioLevel()
-    : abs_max_(0), count_(0), current_level_(0), current_level_full_range_(0) {
+    : abs_max_(0), count_(0), current_level_full_range_(0) {
   WebRtcSpl_Init();
 }
 
 AudioLevel::~AudioLevel() {}
 
-int8_t AudioLevel::Level() const {
-  rtc::CritScope cs(&crit_sect_);
-  return current_level_;
-}
-
 int16_t AudioLevel::LevelFullRange() const {
   rtc::CritScope cs(&crit_sect_);
   return current_level_full_range_;
@@ -44,7 +32,6 @@ void AudioLevel::Clear() {
   rtc::CritScope cs(&crit_sect_);
   abs_max_ = 0;
   count_ = 0;
-  current_level_ = 0;
   current_level_full_range_ = 0;
 }
 
@@ -78,18 +65,6 @@ void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
 
     count_ = 0;
 
-    // Highest value for a int16_t is 0x7fff = 32767
-    // Divide with 1000 to get in the range of 0-32 which is the range of the
-    // permutation vector
-    int32_t position = abs_max_ / 1000;
-
-    // Make it less likely that the bar stays at position 0. I.e. only if it's
-    // in the range 0-250 (instead of 0-1000)
-    if ((position == 0) && (abs_max_ > 250)) {
-      position = 1;
-    }
-    current_level_ = kPermutation[position];
-
     // Decay the absolute maximum (divide by 4)
     abs_max_ >>= 2;
   }

audio/audio_level.h

@@ -25,8 +25,7 @@ class AudioLevel {
   ~AudioLevel();
 
   // Called on "API thread(s)" from APIs like VoEBase::CreateChannel(),
-  // VoEBase::StopSend(), VoEVolumeControl::GetSpeechOutputLevel().
+  // VoEBase::StopSend()
-  int8_t Level() const;
   int16_t LevelFullRange() const;
   void Clear();
   // See the description for "totalAudioEnergy" in the WebRTC stats spec
@@ -46,7 +45,6 @@ class AudioLevel {
 
   int16_t abs_max_ RTC_GUARDED_BY(crit_sect_);
   int16_t count_ RTC_GUARDED_BY(crit_sect_);
-  int8_t current_level_ RTC_GUARDED_BY(crit_sect_);
   int16_t current_level_full_range_ RTC_GUARDED_BY(crit_sect_);
 
   double total_energy_ RTC_GUARDED_BY(crit_sect_) = 0.0;

audio/audio_receive_stream.cc

@@ -221,11 +221,6 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return stats;
 }
 
-int AudioReceiveStream::GetOutputLevel() const {
-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  return channel_proxy_->GetSpeechOutputLevel();
-}
-
 void AudioReceiveStream::SetSink(AudioSinkInterface* sink) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_proxy_->SetSink(sink);

audio/audio_receive_stream.h

@@ -61,7 +61,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void Start() override;
   void Stop() override;
   webrtc::AudioReceiveStream::Stats GetStats() const override;
-  int GetOutputLevel() const override;
   void SetSink(AudioSinkInterface* sink) override;
   void SetGain(float gain) override;
   std::vector<webrtc::RtpSource> GetSources() const override;

audio/audio_state.cc

@@ -149,9 +149,6 @@ AudioState::Stats AudioState::GetAudioInputStats() const {
   result.audio_level = audio_level.LevelFullRange();
   RTC_DCHECK_LE(0, result.audio_level);
   RTC_DCHECK_GE(32767, result.audio_level);
-  result.quantized_audio_level = audio_level.Level();
-  RTC_DCHECK_LE(0, result.quantized_audio_level);
-  RTC_DCHECK_GE(9, result.quantized_audio_level);
   result.total_energy = audio_level.TotalEnergy();
   result.total_duration = audio_level.TotalDuration();
   return result;

audio/audio_state_unittest.cc

@@ -234,7 +234,6 @@ TEST(AudioStateTest, InputLevelStats) {
         kNumChannels, kSampleRate, 0, 0, 0, false, new_mic_level);
     auto stats = audio_state->GetAudioInputStats();
     EXPECT_EQ(0, stats.audio_level);
-    EXPECT_EQ(0, stats.quantized_audio_level);
     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.0));
     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.01));
   }
@@ -250,7 +249,6 @@ TEST(AudioStateTest, InputLevelStats) {
     }
     auto stats = audio_state->GetAudioInputStats();
     EXPECT_EQ(32767, stats.audio_level);
-    EXPECT_EQ(9, stats.quantized_audio_level);
     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.01));
     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.11));
   }

audio/channel.cc

@@ -1011,10 +1011,6 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   return 0;
 }
 
-int Channel::GetSpeechOutputLevel() const {
-  return _outputAudioLevel.Level();
-}
-
 int Channel::GetSpeechOutputLevelFullRange() const {
   return _outputAudioLevel.LevelFullRange();
 }

audio/channel.h

@@ -34,6 +34,10 @@
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
 
+// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
+// warnings about use of unsigned short, and non-const reference arguments.
+// These need cleanup, in a separate cl.
+
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
@@ -57,7 +61,7 @@ class TelephoneEventHandler;
 struct SenderInfo;
 
 struct CallStatistics {
-  unsigned short fractionLost;
+  unsigned short fractionLost;  // NOLINT
   unsigned int cumulativeLost;
   unsigned int extendedMax;
   unsigned int jitterSamples;
@@ -174,7 +178,7 @@ class Channel
   void StopSend();
 
   // Codecs
-  int32_t GetRecCodec(CodecInst& codec);
+  int32_t GetRecCodec(CodecInst& codec);  // NOLINT
   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
   bool EnableAudioNetworkAdaptor(const std::string& config_string);
   void DisableAudioNetworkAdaptor();
@@ -190,7 +194,6 @@ class Channel
   // Muting, Volume and Level.
   void SetInputMute(bool enable);
   void SetChannelOutputVolumeScaling(float scaling);
-  int GetSpeechOutputLevel() const;
   int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec:
   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
@@ -198,14 +201,14 @@ class Channel
   double GetTotalOutputDuration() const;
 
   // Stats.
-  int GetNetworkStatistics(NetworkStatistics& stats);
+  int GetNetworkStatistics(NetworkStatistics& stats);  // NOLINT
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
   ANAStats GetANAStatistics() const;
 
   // Audio+Video Sync.
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
-  int GetPlayoutTimestamp(unsigned int& timestamp);
+  int GetPlayoutTimestamp(unsigned int& timestamp);  // NOLINT
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
   // DTMF.
@@ -226,7 +229,7 @@ class Channel
   void SetRTCPStatus(bool enable);
   int SetRTCP_CNAME(const char cName[256]);
   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
-  int GetRTPStatistics(CallStatistics& stats);
+  int GetRTPStatistics(CallStatistics& stats);  // NOLINT
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
 
   // From AudioPacketizationCallback in the ACM
@@ -265,7 +268,6 @@ class Channel
   bool Playing() const { return channel_state_.Get().playing; }
   bool Sending() const { return channel_state_.Get().sending; }
   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
-  int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
 
   // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
   // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
@@ -309,7 +311,7 @@ class Channel
   void Init();
   void Terminate();
 
-  int GetRemoteSSRC(unsigned int& ssrc);
+  int GetRemoteSSRC(unsigned int& ssrc);  // NOLINT
   void OnUplinkPacketLossRate(float packet_loss_rate);
   bool InputMute() const;
 

audio/channel_proxy.cc

@@ -137,11 +137,6 @@ ANAStats ChannelProxy::GetANAStatistics() const {
   return channel_->GetANAStatistics();
 }
 
-int ChannelProxy::GetSpeechOutputLevel() const {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return channel_->GetSpeechOutputLevel();
-}
-
 int ChannelProxy::GetSpeechOutputLevelFullRange() const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return channel_->GetSpeechOutputLevelFullRange();

audio/channel_proxy.h

@@ -78,7 +78,6 @@ class ChannelProxy : public RtpPacketSinkInterface {
   virtual NetworkStatistics GetNetworkStatistics() const;
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual ANAStats GetANAStatistics() const;
-  virtual int GetSpeechOutputLevel() const;
   virtual int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec:
   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy

audio/mock_voe_channel_proxy.h

@@ -54,7 +54,6 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
   MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
   MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
-  MOCK_CONST_METHOD0(GetSpeechOutputLevel, int());
   MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
   MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
   MOCK_CONST_METHOD0(GetTotalOutputDuration, double());

call/audio_receive_stream.h

@@ -124,8 +124,6 @@ class AudioReceiveStream {
   virtual void Stop() = 0;
 
   virtual Stats GetStats() const = 0;
-  // TODO(solenberg): Remove, once AudioMonitor is gone.
-  virtual int GetOutputLevel() const = 0;
 
   // Sets an audio sink that receives unmixed audio from the receive stream.
   // Ownership of the sink is managed by the caller.

call/audio_state.h

@@ -39,8 +39,6 @@ class AudioState : public rtc::RefCountInterface {
   struct Stats {
     // Audio peak level (max(abs())), linearly on the interval [0,32767].
     int32_t audio_level = -1;
-    // Audio peak level (max(abs())), logarithmically on the interval [0,9].
-    int8_t quantized_audio_level = -1;
     // See: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
     double total_energy = 0.0f;
     double total_duration = 0.0f;

media/base/fakemediaengine.h

@@ -382,9 +382,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
     return true;
   }
 
-  virtual bool GetActiveStreams(StreamList* streams) { return true; }
-  virtual int GetOutputLevel() { return 0; }
-
   virtual bool CanInsertDtmf() {
     for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
          it != send_codecs_.end(); ++it) {

media/base/mediachannel.h

@@ -670,11 +670,6 @@ class VoiceMediaChannel : public MediaChannel {
                             bool enable,
                             const AudioOptions* options,
                             AudioSource* source) = 0;
-  // Gets current energy levels for all incoming streams.
-  typedef std::vector<std::pair<uint32_t, int>> StreamList;
-  virtual bool GetActiveStreams(StreamList* actives) = 0;
-  // Get the current energy level of the stream sent to the speaker.
-  virtual int GetOutputLevel() = 0;
   // Set speaker output volume of the specified ssrc.
   virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
   // Returns if the telephone-event has been negotiated.

media/base/mediaengine.h

@@ -68,9 +68,6 @@ class MediaEngineInterface {
       const MediaConfig& config,
       const VideoOptions& options) = 0;
 
-  // Gets the current microphone level, as a value between 0 and 10.
-  virtual int GetInputLevel() = 0;
-
   virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
   virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
   virtual RtpCapabilities GetAudioCapabilities() = 0;
@@ -119,7 +116,6 @@ class CompositeMediaEngine : public MediaEngineInterface {
     return video().CreateChannel(call, config, options);
   }
 
-  virtual int GetInputLevel() { return voice().GetInputLevel(); }
   virtual const std::vector<AudioCodec>& audio_send_codecs() {
     return voice().send_codecs();
   }

media/engine/fakewebrtccall.h

@@ -100,7 +100,6 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   void Stop() override { started_ = false; }
 
   webrtc::AudioReceiveStream::Stats GetStats() const override;
-  int GetOutputLevel() const override { return 0; }
   void SetSink(webrtc::AudioSinkInterface* sink) override;
   void SetGain(float gain) override;
   std::vector<webrtc::RtpSource> GetSources() const override {

media/engine/webrtcvoiceengine.cc

@@ -593,12 +593,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   return true;
 }
 
-// TODO(solenberg): Remove, once AudioMonitor is gone.
-int WebRtcVoiceEngine::GetInputLevel() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return audio_state()->GetAudioInputStats().quantized_audio_level;
-}
-
 const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   return send_codecs_;
@@ -1193,12 +1187,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     return stream_->GetStats();
   }
 
-  int GetOutputLevel() const {
-    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    RTC_DCHECK(stream_);
-    return stream_->GetOutputLevel();
-  }
-
   void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     raw_audio_sink_ = std::move(sink);
@@ -1923,30 +1911,6 @@ bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc,
   return true;
 }
 
-// TODO(solenberg): Remove, once AudioMonitor is gone.
-bool WebRtcVoiceMediaChannel::GetActiveStreams(
-    AudioInfo::StreamList* actives) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  actives->clear();
-  for (const auto& ch : recv_streams_) {
-    int level = ch.second->GetOutputLevel();
-    if (level > 0) {
-      actives->push_back(std::make_pair(ch.first, level));
-    }
-  }
-  return true;
-}
-
-// TODO(solenberg): Remove, once AudioMonitor is gone.
-int WebRtcVoiceMediaChannel::GetOutputLevel() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  int highest = 0;
-  for (const auto& ch : recv_streams_) {
-    highest = std::max(ch.second->GetOutputLevel(), highest);
-  }
-  return highest;
-}
-
 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   std::vector<uint32_t> ssrcs(1, ssrc);

media/engine/webrtcvoiceengine.h

@@ -59,8 +59,6 @@ class WebRtcVoiceEngine final {
                                    const MediaConfig& config,
                                    const AudioOptions& options);
 
-  int GetInputLevel();
-
   const std::vector<AudioCodec>& send_codecs() const;
   const std::vector<AudioCodec>& recv_codecs() const;
   RtpCapabilities GetCapabilities() const;
@@ -174,8 +172,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool RemoveSendStream(uint32_t ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
   bool RemoveRecvStream(uint32_t ssrc) override;
-  bool GetActiveStreams(StreamList* actives) override;
-  int GetOutputLevel() override;
   // SSRC=0 will apply the new volume to current and future unsignaled streams.
   bool SetOutputVolume(uint32_t ssrc, double volume) override;
 

pc/BUILD.gn

@@ -29,7 +29,6 @@ rtc_static_library("rtc_pc_base") {
   visibility = [ "*" ]
   defines = []
   sources = [
-    "audiomonitor.cc",
     "audiomonitor.h",
     "bundlefilter.cc",
     "bundlefilter.h",
@@ -65,7 +64,6 @@ rtc_static_library("rtc_pc_base") {
     "srtptransport.h",
     "transportcontroller.cc",
     "transportcontroller.h",
-    "voicechannel.h",
   ]
 
   deps = [

pc/audiomonitor.cc

@@ -1,105 +0,0 @@
-/*
- *  Copyright 2004 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 "pc/audiomonitor.h"
-#include "pc/voicechannel.h"
-
-namespace cricket {
-
-const uint32_t MSG_MONITOR_POLL = 1;
-const uint32_t MSG_MONITOR_START = 2;
-const uint32_t MSG_MONITOR_STOP = 3;
-const uint32_t MSG_MONITOR_SIGNAL = 4;
-
-AudioMonitor::AudioMonitor(VoiceChannel *voice_channel,
-                           rtc::Thread *monitor_thread) {
-  voice_channel_ = voice_channel;
-  monitoring_thread_ = monitor_thread;
-  monitoring_ = false;
-}
-
-AudioMonitor::~AudioMonitor() {
-  voice_channel_->worker_thread()->Clear(this);
-  monitoring_thread_->Clear(this);
-}
-
-void AudioMonitor::Start(int milliseconds) {
-  rate_ = milliseconds;
-  if (rate_ < 100)
-    rate_ = 100;
-  voice_channel_->worker_thread()->Post(RTC_FROM_HERE, this, MSG_MONITOR_START);
-}
-
-void AudioMonitor::Stop() {
-  voice_channel_->worker_thread()->Post(RTC_FROM_HERE, this, MSG_MONITOR_STOP);
-}
-
-void AudioMonitor::OnMessage(rtc::Message *message) {
-  rtc::CritScope cs(&crit_);
-
-  switch (message->message_id) {
-  case MSG_MONITOR_START:
-    assert(rtc::Thread::Current() == voice_channel_->worker_thread());
-    if (!monitoring_) {
-      monitoring_ = true;
-      PollVoiceChannel();
-    }
-    break;
-
-  case MSG_MONITOR_STOP:
-    assert(rtc::Thread::Current() == voice_channel_->worker_thread());
-    if (monitoring_) {
-      monitoring_ = false;
-      voice_channel_->worker_thread()->Clear(this);
-    }
-    break;
-
-  case MSG_MONITOR_POLL:
-    assert(rtc::Thread::Current() == voice_channel_->worker_thread());
-    PollVoiceChannel();
-    break;
-
-  case MSG_MONITOR_SIGNAL:
-    {
-      assert(rtc::Thread::Current() == monitoring_thread_);
-      AudioInfo info = audio_info_;
-      crit_.Leave();
-      SignalUpdate(this, info);
-      crit_.Enter();
-    }
-    break;
-  }
-}
-
-void AudioMonitor::PollVoiceChannel() {
-  rtc::CritScope cs(&crit_);
-  assert(rtc::Thread::Current() == voice_channel_->worker_thread());
-
-  // Gather connection infos
-  audio_info_.input_level = voice_channel_->GetInputLevel_w();
-  audio_info_.output_level = voice_channel_->GetOutputLevel_w();
-  voice_channel_->GetActiveStreams_w(&audio_info_.active_streams);
-
-  // Signal the monitoring thread, start another poll timer
-  monitoring_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_SIGNAL);
-  voice_channel_->worker_thread()->PostDelayed(RTC_FROM_HERE, rate_, this,
-                                               MSG_MONITOR_POLL);
-}
-
-VoiceChannel *AudioMonitor::voice_channel() {
-  return voice_channel_;
-}
-
-rtc::Thread *AudioMonitor::monitor_thread() {
-  return monitoring_thread_;
-}
-
-}  // namespace cricket

pc/audiomonitor.h

@@ -14,14 +14,11 @@
 #include <vector>
 #include <utility>
 
+// For ConnectionInfo/ConnectionInfos
 #include "p2p/base/port.h"
-#include "rtc_base/sigslot.h"
-#include "rtc_base/thread.h"
 
 namespace cricket {
 
-class VoiceChannel;
-
 struct AudioInfo {
   int input_level;
   int output_level;
@@ -29,32 +26,6 @@ struct AudioInfo {
   StreamList active_streams;  // ssrcs contributing to output_level
 };
 
-class AudioMonitor : public rtc::MessageHandler,
-    public sigslot::has_slots<> {
- public:
-  AudioMonitor(VoiceChannel* voice_channel, rtc::Thread *monitor_thread);
-  ~AudioMonitor();
-
-  void Start(int cms);
-  void Stop();
-
-  VoiceChannel* voice_channel();
-  rtc::Thread *monitor_thread();
-
-  sigslot::signal2<AudioMonitor*, const AudioInfo&> SignalUpdate;
-
- protected:
-  void OnMessage(rtc::Message *message);
-  void PollVoiceChannel();
-
-  AudioInfo audio_info_;
-  VoiceChannel* voice_channel_;
-  rtc::Thread* monitoring_thread_;
-  rtc::CriticalSection crit_;
-  uint32_t rate_;
-  bool monitoring_;
-};
-
 }  // namespace cricket
 
 #endif  // PC_AUDIOMONITOR_H_

pc/channel.cc

@@ -1134,7 +1134,8 @@ void BaseChannel::ActivateRtcpMux() {
 VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
                            rtc::Thread* network_thread,
                            rtc::Thread* signaling_thread,
-                           MediaEngineInterface* media_engine,
+                           // TODO(nisse): Delete unused argument.
+                           MediaEngineInterface* /* media_engine */,
                            std::unique_ptr<VoiceMediaChannel> media_channel,
                            const std::string& content_name,
                            bool rtcp_mux_required,
@@ -1145,12 +1146,10 @@ VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
                   std::move(media_channel),
                   content_name,
                   rtcp_mux_required,
-                  srtp_required),
+                  srtp_required) {}
-      media_engine_(media_engine) {}
 
 VoiceChannel::~VoiceChannel() {
   TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
-  StopAudioMonitor();
   // this can't be done in the base class, since it calls a virtual
   DisableMedia_w();
   Deinit();
@@ -1186,34 +1185,6 @@ bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
                                                   media_channel(), stats));
 }
 
-void VoiceChannel::StartAudioMonitor(int cms) {
-  audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current()));
-  audio_monitor_->Start(cms);
-}
-
-void VoiceChannel::StopAudioMonitor() {
-  if (audio_monitor_) {
-    audio_monitor_->Stop();
-    audio_monitor_.reset();
-  }
-}
-
-bool VoiceChannel::IsAudioMonitorRunning() const {
-  return (audio_monitor_.get() != NULL);
-}
-
-int VoiceChannel::GetInputLevel_w() {
-  return media_engine_->GetInputLevel();
-}
-
-int VoiceChannel::GetOutputLevel_w() {
-  return media_channel()->GetOutputLevel();
-}
-
-void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
-  media_channel()->GetActiveStreams(actives);
-}
-
 void VoiceChannel::OnPacketReceived(bool rtcp,
                                     rtc::CopyOnWriteBuffer* packet,
                                     const rtc::PacketTime& packet_time) {

pc/channel.h

@@ -471,14 +471,6 @@ class VoiceChannel : public BaseChannel {
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
-  // Monitoring functions
-  void StartAudioMonitor(int cms);
-  void StopAudioMonitor();
-  bool IsAudioMonitorRunning() const;
-
-  int GetInputLevel_w();
-  int GetOutputLevel_w();
-  void GetActiveStreams_w(AudioInfo::StreamList* actives);
   webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
   webrtc::RTCError SetRtpSendParameters_w(uint32_t ssrc,
                                           webrtc::RtpParameters parameters);
@@ -501,9 +493,7 @@ class VoiceChannel : public BaseChannel {
   void OnMessage(rtc::Message* pmsg) override;
 
   static const int kEarlyMediaTimeout = 1000;
-  MediaEngineInterface* media_engine_;
   bool received_media_ = false;
-  std::unique_ptr<AudioMonitor> audio_monitor_;
 
   // Last AudioSendParameters sent down to the media_channel() via
   // SetSendParameters.

pc/currentspeakermonitor.cc

@@ -15,6 +15,7 @@
 #include "media/base/streamparams.h"
 #include "pc/audiomonitor.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/timeutils.h"
 
 namespace cricket {
 

pc/voicechannel.h

@@ -1,16 +0,0 @@
-/*
- *  Copyright 2004 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 PC_VOICECHANNEL_H_
-#define PC_VOICECHANNEL_H_
-
-#include "pc/channel.h"
-
-#endif  // PC_VOICECHANNEL_H_