Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"

This reverts commit 90bace0958.

Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.

Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
> 
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
> 
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
> 
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
> 
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
> 
> TBR=solenberg
> 
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}

TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org

Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}

api/peerconnectioninterface.h

@@ -788,21 +788,6 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
       std::unique_ptr<rtc::BitrateAllocationStrategy>
           bitrate_allocation_strategy) {}
 
-  // Enable/disable playout of received audio streams. Enabled by default. Note
-  // that even if playout is enabled, streams will only be played out if the
-  // appropriate SDP is also applied. Setting |playout| to false will stop
-  // playout of the underlying audio device but starts a task which will poll
-  // for audio data every 10ms to ensure that audio processing happens and the
-  // audio statistics are updated.
-  // TODO(henrika): deprecate and remove this.
-  virtual void SetAudioPlayout(bool playout) {}
-
-  // Enable/disable recording of transmitted audio streams. Enabled by default.
-  // Note that even if recording is enabled, streams will only be recorded if
-  // the appropriate SDP is also applied.
-  // TODO(henrika): deprecate and remove this.
-  virtual void SetAudioRecording(bool recording) {}
-
   // Returns the current SignalingState.
   virtual SignalingState signaling_state() = 0;
 

api/peerconnectionproxy.h

@@ -100,8 +100,6 @@ BEGIN_SIGNALING_PROXY_MAP(PeerConnection)
   PROXY_METHOD1(bool,
                 RemoveIceCandidates,
                 const std::vector<cricket::Candidate>&);
-  PROXY_METHOD1(void, SetAudioPlayout, bool)
-  PROXY_METHOD1(void, SetAudioRecording, bool)
   PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*)
   PROXY_METHOD1(RTCError, SetBitrate, const BitrateParameters&);
   PROXY_METHOD1(void,

audio/BUILD.gn

@@ -23,8 +23,6 @@ rtc_static_library("audio") {
     "audio_transport_proxy.cc",
     "audio_transport_proxy.h",
     "conversion.h",
-    "null_audio_poller.cc",
-    "null_audio_poller.h",
     "scoped_voe_interface.h",
     "time_interval.cc",
     "time_interval.h",
@@ -54,7 +52,6 @@ rtc_static_library("audio") {
     "../modules/pacing:pacing",
     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
     "../modules/rtp_rtcp:rtp_rtcp",
-    "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:rtc_task_queue",
     "../system_wrappers",

audio/audio_state.cc

@@ -12,11 +12,8 @@
 
 #include "modules/audio_device/include/audio_device.h"
 #include "rtc_base/atomicops.h"
-#include "rtc_base/bind.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/ptr_util.h"
-#include "rtc_base/thread.h"
 #include "voice_engine/transmit_mixer.h"
 
 namespace webrtc {
@@ -62,40 +59,6 @@ bool AudioState::typing_noise_detected() const {
   return transmit_mixer->typing_noise_detected();
 }
 
-void AudioState::SetPlayout(bool enabled) {
-  LOG(INFO) << "SetPlayout(" << enabled << ")";
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  const bool currently_enabled = (null_audio_poller_ == nullptr);
-  if (enabled == currently_enabled) {
-    return;
-  }
-  VoEBase* const voe = VoEBase::GetInterface(voice_engine());
-  RTC_DCHECK(voe);
-  if (enabled) {
-    null_audio_poller_.reset();
-  }
-  // Will stop/start playout of the underlying device, if necessary, and
-  // remember the setting for when it receives subsequent calls of
-  // StartPlayout.
-  voe->SetPlayout(enabled);
-  if (!enabled) {
-    null_audio_poller_ =
-        rtc::MakeUnique<NullAudioPoller>(&audio_transport_proxy_);
-  }
-}
-
-void AudioState::SetRecording(bool enabled) {
-  LOG(INFO) << "SetRecording(" << enabled << ")";
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  // TODO(henrika): keep track of state as in SetPlayout().
-  VoEBase* const voe = VoEBase::GetInterface(voice_engine());
-  RTC_DCHECK(voe);
-  // Will stop/start recording of the underlying device, if necessary, and
-  // remember the setting for when it receives subsequent calls of
-  // StartPlayout.
-  voe->SetRecording(enabled);
-}
-
 // Reference count; implementation copied from rtc::RefCountedObject.
 void AudioState::AddRef() const {
   rtc::AtomicOps::Increment(&ref_count_);

audio/audio_state.h

@@ -11,10 +11,7 @@
 #ifndef AUDIO_AUDIO_STATE_H_
 #define AUDIO_AUDIO_STATE_H_
 
-#include <memory>
-
 #include "audio/audio_transport_proxy.h"
-#include "audio/null_audio_poller.h"
 #include "audio/scoped_voe_interface.h"
 #include "call/audio_state.h"
 #include "rtc_base/constructormagic.h"
@@ -36,9 +33,6 @@ class AudioState final : public webrtc::AudioState {
     return config_.audio_processing.get();
   }
 
-  void SetPlayout(bool enabled) override;
-  void SetRecording(bool enabled) override;
-
   VoiceEngine* voice_engine();
   rtc::scoped_refptr<AudioMixer> mixer();
   bool typing_noise_detected() const;
@@ -63,11 +57,6 @@ class AudioState final : public webrtc::AudioState {
   // recorded audio to the VoE AudioTransport.
   AudioTransportProxy audio_transport_proxy_;
 
-  // Null audio poller is used to continue polling the audio streams if audio
-  // playout is disabled so that audio processing still happens and the audio
-  // stats are still updated.
-  std::unique_ptr<NullAudioPoller> null_audio_poller_;
-
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
 };
 }  // namespace internal

audio/null_audio_poller.cc

@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2017 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 "audio/null_audio_poller.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/thread.h"
-
-namespace webrtc {
-namespace internal {
-
-namespace {
-
-constexpr int64_t kPollDelayMs = 10;  // WebRTC uses 10ms by default
-
-constexpr size_t kNumChannels = 1;
-constexpr uint32_t kSamplesPerSecond = 48000;            // 48kHz
-constexpr size_t kNumSamples = kSamplesPerSecond / 100;  // 10ms of samples
-
-}  // namespace
-
-NullAudioPoller::NullAudioPoller(AudioTransport* audio_transport)
-    : audio_transport_(audio_transport),
-      reschedule_at_(rtc::TimeMillis() + kPollDelayMs) {
-  RTC_DCHECK(audio_transport);
-  OnMessage(nullptr);  // Start the poll loop.
-}
-
-NullAudioPoller::~NullAudioPoller() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  rtc::Thread::Current()->Clear(this);
-}
-
-void NullAudioPoller::OnMessage(rtc::Message* msg) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-
-  // Buffer to hold the audio samples.
-  int16_t buffer[kNumSamples * kNumChannels];
-  // Output variables from |NeedMorePlayData|.
-  size_t n_samples;
-  int64_t elapsed_time_ms;
-  int64_t ntp_time_ms;
-  audio_transport_->NeedMorePlayData(kNumSamples, sizeof(int16_t), kNumChannels,
-                                     kSamplesPerSecond, buffer, n_samples,
-                                     &elapsed_time_ms, &ntp_time_ms);
-
-  // Reschedule the next poll iteration. If, for some reason, the given
-  // reschedule time has already passed, reschedule as soon as possible.
-  int64_t now = rtc::TimeMillis();
-  if (reschedule_at_ < now) {
-    reschedule_at_ = now;
-  }
-  rtc::Thread::Current()->PostAt(RTC_FROM_HERE, reschedule_at_, this, 0);
-
-  // Loop after next will be kPollDelayMs later.
-  reschedule_at_ += kPollDelayMs;
-}
-
-}  // namespace internal
-}  // namespace webrtc

audio/null_audio_poller.h

@@ -1,38 +0,0 @@
-/*
- *  Copyright (c) 2017 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 AUDIO_NULL_AUDIO_POLLER_H_
-#define AUDIO_NULL_AUDIO_POLLER_H_
-
-#include "modules/audio_device/include/audio_device_defines.h"
-#include "rtc_base/messagehandler.h"
-#include "rtc_base/thread_checker.h"
-
-namespace webrtc {
-namespace internal {
-
-class NullAudioPoller final : public rtc::MessageHandler {
- public:
-  explicit NullAudioPoller(AudioTransport* audio_transport);
-  ~NullAudioPoller();
-
- protected:
-  void OnMessage(rtc::Message* msg) override;
-
- private:
-  const rtc::ThreadChecker thread_checker_;
-  AudioTransport* const audio_transport_;
-  int64_t reschedule_at_;
-};
-
-}  // namespace internal
-}  // namespace webrtc
-
-#endif  // AUDIO_NULL_AUDIO_POLLER_H_

call/audio_state.h

@@ -44,17 +44,6 @@ class AudioState : public rtc::RefCountInterface {
 
   virtual AudioProcessing* audio_processing() = 0;
 
-  // Enable/disable playout of the audio channels. Enabled by default.
-  // This will stop playout of the underlying audio device but start a task
-  // which will poll for audio data every 10ms to ensure that audio processing
-  // happens and the audio stats are updated.
-  virtual void SetPlayout(bool enabled) = 0;
-
-  // Enable/disable recording of the audio channels. Enabled by default.
-  // This will stop recording of the underlying audio device and no audio
-  // packets will be encoded or transmitted.
-  virtual void SetRecording(bool enabled) = 0;
-
   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
   static rtc::scoped_refptr<AudioState> Create(
       const AudioState::Config& config);

media/engine/fakewebrtcvoiceengine.h

@@ -99,8 +99,6 @@ class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
   WEBRTC_STUB(StartSend, (int channel));
   WEBRTC_STUB(StopPlayout, (int channel));
   WEBRTC_STUB(StopSend, (int channel));
-  WEBRTC_STUB(SetPlayout, (bool enable));
-  WEBRTC_STUB(SetRecording, (bool enable));
 
   size_t GetNetEqCapacity() const {
     auto ch = channels_.find(last_channel_);

pc/BUILD.gn

@@ -183,7 +183,6 @@ rtc_static_library("peerconnection") {
     "../rtc_base:rtc_base_approved",
     "../stats",
     "../system_wrappers:system_wrappers",
-    "../voice_engine:voice_engine",
   ]
 
   public_deps = [

pc/peerconnection.cc

@@ -1323,30 +1323,6 @@ void PeerConnection::SetBitrateAllocationStrategy(
   call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
 }
 
-void PeerConnection::SetAudioPlayout(bool playout) {
-  if (!worker_thread()->IsCurrent()) {
-    worker_thread()->Invoke<void>(
-        RTC_FROM_HERE,
-        rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
-    return;
-  }
-  auto audio_state =
-      factory_->channel_manager()->media_engine()->GetAudioState();
-  audio_state->SetPlayout(playout);
-}
-
-void PeerConnection::SetAudioRecording(bool recording) {
-  if (!worker_thread()->IsCurrent()) {
-    worker_thread()->Invoke<void>(
-        RTC_FROM_HERE,
-        rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
-    return;
-  }
-  auto audio_state =
-      factory_->channel_manager()->media_engine()->GetAudioState();
-  audio_state->SetRecording(recording);
-}
-
 std::unique_ptr<rtc::SSLCertificate>
 PeerConnection::GetRemoteAudioSSLCertificate() {
   if (!session_) {

pc/peerconnection.h

@@ -143,9 +143,6 @@ class PeerConnection : public PeerConnectionInterface,
       std::unique_ptr<rtc::BitrateAllocationStrategy>
           bitrate_allocation_strategy) override;
 
-  void SetAudioPlayout(bool playout) override;
-  void SetAudioRecording(bool recording) override;
-
   RTC_DEPRECATED bool StartRtcEventLog(rtc::PlatformFile file,
                                        int64_t max_size_bytes) override;
   bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override;

pc/peerconnection_integrationtest.cc

@@ -3564,76 +3564,6 @@ TEST_F(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
       kMaxWaitForFramesMs);
 }
 
-// Test that SetAudioPlayout can be used to disable audio playout from the
-// start, then later enable it. This may be useful, for example, if the caller
-// needs to play a local ringtone until some event occurs, after which it
-// switches to playing the received audio.
-TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
-  ASSERT_TRUE(CreatePeerConnectionWrappers());
-  ConnectFakeSignaling();
-
-  // Set up audio-only call where audio playout is disabled on caller's side.
-  caller()->pc()->SetAudioPlayout(false);
-  caller()->AddAudioOnlyMediaStream();
-  callee()->AddAudioOnlyMediaStream();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
-  // Pump messages for a second.
-  WAIT(false, 1000);
-  // Since audio playout is disabled, the caller shouldn't have received
-  // anything (at the playout level, at least).
-  EXPECT_EQ(0, caller()->audio_frames_received());
-  // As a sanity check, make sure the callee (for which playout isn't disabled)
-  // did still see frames on its audio level.
-  ASSERT_GT(callee()->audio_frames_received(), 0);
-
-  // Enable playout again, and ensure audio starts flowing.
-  caller()->pc()->SetAudioPlayout(true);
-  ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0,
-                                  kDefaultExpectedAudioFrameCount, 0,
-                                  kMaxWaitForFramesMs);
-}
-
-double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
-  auto report = pc->NewGetStats();
-  auto track_stats_list =
-      report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
-  const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr;
-  for (const auto* track_stats : track_stats_list) {
-    if (track_stats->remote_source.is_defined() &&
-        *track_stats->remote_source) {
-      remote_track_stats = track_stats;
-      break;
-    }
-  }
-
-  if (!remote_track_stats->total_audio_energy.is_defined()) {
-    return 0.0;
-  }
-  return *remote_track_stats->total_audio_energy;
-}
-
-// Test that if audio playout is disabled via the SetAudioPlayout() method, then
-// incoming audio is still processed and statistics are generated.
-TEST_F(PeerConnectionIntegrationTest,
-       DisableAudioPlayoutStillGeneratesAudioStats) {
-  ASSERT_TRUE(CreatePeerConnectionWrappers());
-  ConnectFakeSignaling();
-
-  // Set up audio-only call where playout is disabled but audio-processing is
-  // still active.
-  caller()->AddAudioOnlyMediaStream();
-  callee()->AddAudioOnlyMediaStream();
-  caller()->pc()->SetAudioPlayout(false);
-
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
-  // Wait for the callee to receive audio stats.
-  EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
-}
-
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)

sdk/android/api/org/webrtc/PeerConnection.java

@@ -363,18 +363,6 @@ public class PeerConnection {
 
   public native void setRemoteDescription(SdpObserver observer, SessionDescription sdp);
 
-  // True if remote audio should be played out. Defaults to true.
-  // Note that even if playout is enabled, streams will only be played out if
-  // the appropriate SDP is also applied. The main purpose of this API is to
-  // be able to control the exact time when audio playout starts.
-  public native void setAudioPlayout(boolean playout);
-
-  // True if local audio shall be recorded. Defaults to true.
-  // Note that even if recording is enabled, streams will only be recorded if
-  // the appropriate SDP is also applied. The main purpose of this API is to
-  // be able to control the exact time when audio recording starts.
-  public native void setAudioRecording(boolean recording);
-
   public boolean setConfiguration(RTCConfiguration config) {
     return nativeSetConfiguration(config, nativeObserver);
   }

sdk/android/src/jni/pc/peerconnection_jni.cc

@@ -166,22 +166,6 @@ JNI_FUNCTION_DECLARATION(void,
       observer, JavaToNativeSessionDescription(jni, j_sdp));
 }
 
-JNI_FUNCTION_DECLARATION(void,
-                         PeerConnection_setAudioPlayout,
-                         JNIEnv* jni,
-                         jobject j_pc,
-                         jboolean playout) {
-  ExtractNativePC(jni, j_pc)->SetAudioPlayout(playout);
-}
-
-JNI_FUNCTION_DECLARATION(void,
-                         PeerConnection_setAudioRecording,
-                         JNIEnv* jni,
-                         jobject j_pc,
-                         jboolean recording) {
-  ExtractNativePC(jni, j_pc)->SetAudioRecording(recording);
-}
-
 JNI_FUNCTION_DECLARATION(jboolean,
                          PeerConnection_nativeSetConfiguration,
                          JNIEnv* jni,

voice_engine/include/voe_base.h

@@ -139,21 +139,6 @@ class WEBRTC_DLLEXPORT VoEBase {
   // Stops sending packets from a specified |channel|.
   virtual int StopSend(int channel) = 0;
 
-  // Enable or disable playout to the underlying device. Takes precedence over
-  // StartPlayout. Though calls to StartPlayout are remembered; if
-  // SetPlayout(true) is called after StartPlayout, playout will be started.
-  //
-  // By default, playout is enabled.
-  virtual int SetPlayout(bool enabled) = 0;
-
-  // Enable or disable recording (which drives sending of encoded audio packtes)
-  // from the underlying device. Takes precedence over StartSend. Though calls
-  // to StartSend are remembered; if SetRecording(true) is called after
-  // StartSend, recording will be started.
-  //
-  // By default, recording is enabled.
-  virtual int SetRecording(bool enabled) = 0;
-
   // TODO(xians): Make the interface pure virtual after libjingle
   // implements the interface in its FakeWebRtcVoiceEngine.
   virtual AudioTransport* audio_transport() { return NULL; }

voice_engine/voe_base_impl.cc

@@ -407,7 +407,7 @@ int32_t VoEBaseImpl::StartPlayout() {
       LOG_F(LS_ERROR) << "Failed to initialize playout";
       return -1;
     }
-    if (playout_enabled_ && shared_->audio_device()->StartPlayout() != 0) {
+    if (shared_->audio_device()->StartPlayout() != 0) {
       LOG_F(LS_ERROR) << "Failed to start playout";
       return -1;
     }
@@ -416,10 +416,7 @@ int32_t VoEBaseImpl::StartPlayout() {
 }
 
 int32_t VoEBaseImpl::StopPlayout() {
-  if (!playout_enabled_) {
+  // Stop audio-device playing if no channel is playing out
-    return 0;
-  }
-  // Stop audio-device playing if no channel is playing out.
   if (shared_->NumOfPlayingChannels() == 0) {
     if (shared_->audio_device()->StopPlayout() != 0) {
       LOG(LS_ERROR) << "StopPlayout() failed to stop playout";
@@ -430,12 +427,15 @@ int32_t VoEBaseImpl::StopPlayout() {
 }
 
 int32_t VoEBaseImpl::StartSend() {
-  if (!shared_->audio_device()->Recording()) {
+  if (!shared_->audio_device()->RecordingIsInitialized() &&
+      !shared_->audio_device()->Recording()) {
     if (shared_->audio_device()->InitRecording() != 0) {
       LOG_F(LS_ERROR) << "Failed to initialize recording";
       return -1;
     }
-    if (recording_enabled_ && shared_->audio_device()->StartRecording() != 0) {
+  }
+  if (!shared_->audio_device()->Recording()) {
+    if (shared_->audio_device()->StartRecording() != 0) {
       LOG_F(LS_ERROR) << "Failed to start recording";
       return -1;
     }
@@ -444,11 +444,8 @@ int32_t VoEBaseImpl::StartSend() {
 }
 
 int32_t VoEBaseImpl::StopSend() {
-  if (!recording_enabled_) {
-    return 0;
-  }
-  // Stop audio-device recording if no channel is recording.
   if (shared_->NumOfSendingChannels() == 0) {
+    // Stop audio-device recording if no channel is recording
     if (shared_->audio_device()->StopRecording() != 0) {
       LOG(LS_ERROR) << "StopSend() failed to stop recording";
       return -1;
@@ -459,58 +456,6 @@ int32_t VoEBaseImpl::StopSend() {
   return 0;
 }
 
-int32_t VoEBaseImpl::SetPlayout(bool enabled) {
-  LOG(INFO) << "SetPlayout(" << enabled << ")";
-  if (playout_enabled_ == enabled) {
-    return 0;
-  }
-  playout_enabled_ = enabled;
-  if (shared_->NumOfPlayingChannels() == 0) {
-    // If there are no channels attempting to play out yet, there's nothing to
-    // be done; we should be in a "not playing out" state either way.
-    return 0;
-  }
-  int32_t ret;
-  if (enabled) {
-    ret = shared_->audio_device()->StartPlayout();
-    if (ret != 0) {
-      LOG(LS_ERROR) << "SetPlayout(true) failed to start playout";
-    }
-  } else {
-    ret = shared_->audio_device()->StopPlayout();
-    if (ret != 0) {
-      LOG(LS_ERROR) << "SetPlayout(false) failed to stop playout";
-    }
-  }
-  return ret;
-}
-
-int32_t VoEBaseImpl::SetRecording(bool enabled) {
-  LOG(INFO) << "SetRecording(" << enabled << ")";
-  if (recording_enabled_ == enabled) {
-    return 0;
-  }
-  recording_enabled_ = enabled;
-  if (shared_->NumOfSendingChannels() == 0) {
-    // If there are no channels attempting to record out yet, there's nothing to
-    // be done; we should be in a "not recording" state either way.
-    return 0;
-  }
-  int32_t ret;
-  if (enabled) {
-    ret = shared_->audio_device()->StartRecording();
-    if (ret != 0) {
-      LOG(LS_ERROR) << "SetRecording(true) failed to start recording";
-    }
-  } else {
-    ret = shared_->audio_device()->StopRecording();
-    if (ret != 0) {
-      LOG(LS_ERROR) << "SetRecording(false) failed to stop recording";
-    }
-  }
-  return ret;
-}
-
 int32_t VoEBaseImpl::TerminateInternal() {
   // Delete any remaining channel objects
   shared_->channel_manager().DestroyAllChannels();

voice_engine/voe_base_impl.h

@@ -45,9 +45,6 @@ class VoEBaseImpl : public VoEBase,
   int StopPlayout(int channel) override;
   int StopSend(int channel) override;
 
-  int SetPlayout(bool enabled) override;
-  int SetRecording(bool enabled) override;
-
   AudioTransport* audio_transport() override { return this; }
 
   // AudioTransport
@@ -106,8 +103,6 @@ class VoEBaseImpl : public VoEBase,
 
   AudioFrame audioFrame_;
   voe::SharedData* shared_;
-  bool playout_enabled_ = true;
-  bool recording_enabled_ = true;
 };
 
 }  // namespace webrtc