Removes more unused ADM APIs:

- RecordingDelay()
- LastError()

Bug: webrtc:7306
Change-Id: I3bb9cd243a1464f0ba612787c854eeb6602c7e38
Reviewed-on: https://webrtc-review.googlesource.com/12060
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20326}

media/engine/webrtcvoiceengine.cc

@@ -620,8 +620,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
                  << *options.recording_sample_rate;
     if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
       LOG(LS_WARNING) << "SetRecordingSampleRate("
-                      << *options.recording_sample_rate << ") failed, err="
+                      << *options.recording_sample_rate << ") failed.";
-                      << adm()->LastError();
     }
   }
 
@@ -629,8 +628,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
     if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
       LOG(LS_WARNING) << "SetPlayoutSampleRate("
-                      << *options.playout_sample_rate << ") failed, err="
+                      << *options.playout_sample_rate << ") failed.";
-                      << adm()->LastError();
     }
   }
   return true;

modules/audio_device/android/audio_device_template.h

@@ -383,14 +383,6 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
     return 0;
   }
 
-  int32_t RecordingDelay(uint16_t& delay_ms) const override {
-    // Best guess we can do is to use half of the estimated total delay.
-    LOG(INFO) << __FUNCTION__;
-    delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
-    RTC_DCHECK_GT(delay_ms, 0);
-    return 0;
-  }
-
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
     LOG(INFO) << __FUNCTION__;
     output_.AttachAudioBuffer(audioBuffer);

modules/audio_device/audio_device_data_observer.cc

@@ -120,7 +120,6 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
   int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override {
     return impl_->ActiveAudioLayer(audio_layer);
   }
-  ErrorCode LastError() const override { return impl_->LastError(); }
   int32_t Init() override { return impl_->Init(); }
   int32_t Terminate() override { return impl_->Terminate(); }
   bool Initialized() const override { return impl_->Initialized(); }
@@ -253,9 +252,6 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
   int32_t PlayoutDelay(uint16_t* delay_ms) const override {
     return impl_->PlayoutDelay(delay_ms);
   }
-  int32_t RecordingDelay(uint16_t* delay_ms) const override {
-    return impl_->RecordingDelay(delay_ms);
-  }
   int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override {
     return impl_->SetRecordingSampleRate(samples_per_sec);
   }

modules/audio_device/audio_device_generic.h

@@ -115,7 +115,6 @@ class AudioDeviceGeneric {
 
   // Delay information and control
   virtual int32_t PlayoutDelay(uint16_t& delayMS) const = 0;
-  virtual int32_t RecordingDelay(uint16_t& delayMS) const = 0;
 
   // Native sample rate controls (samples/sec)
   virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec);

modules/audio_device/audio_device_impl.cc

@@ -272,13 +272,6 @@ int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
   return 0;
 }
 
-// TODO(henrika): remove this API.
-AudioDeviceModule::ErrorCode AudioDeviceModuleImpl::LastError() const {
-  LOG(INFO) << __FUNCTION__;
-  LOG(WARNING) << "Not supported";
-  return kAdmErrNone;
-}
-
 int32_t AudioDeviceModuleImpl::Init() {
   LOG(INFO) << __FUNCTION__;
   if (initialized_)
@@ -858,19 +851,6 @@ int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const {
   return 0;
 }
 
-int32_t AudioDeviceModuleImpl::RecordingDelay(uint16_t* delayMS) const {
-  LOG(INFO) << __FUNCTION__;
-  CHECKinitialized_();
-  uint16_t delay = 0;
-  if (audio_device_->RecordingDelay(delay) == -1) {
-    LOG(LERROR) << "failed to retrieve the recording delay";
-    return -1;
-  }
-  *delayMS = delay;
-  LOG(INFO) << "output: " << *delayMS;
-  return 0;
-}
-
 int32_t AudioDeviceModuleImpl::SetRecordingSampleRate(
     const uint32_t samplesPerSec) {
   LOG(INFO) << __FUNCTION__ << "(" << samplesPerSec << ")";

modules/audio_device/audio_device_impl.h

@@ -47,9 +47,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
   // Retrieve the currently utilized audio layer
   int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
 
-  // Error handling
-  ErrorCode LastError() const override;
-
   // Full-duplex transportation of PCM audio
   int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
 
@@ -136,7 +133,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
 
   // Delay information and control
   int32_t PlayoutDelay(uint16_t* delayMS) const override;
-  int32_t RecordingDelay(uint16_t* delayMS) const override;
 
   // Native sample rate controls (samples/sec)
   int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override;

modules/audio_device/dummy/audio_device_dummy.cc

@@ -156,7 +156,5 @@ int32_t AudioDeviceDummy::StereoRecording(bool& enabled) const { return -1; }
 
 int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const { return -1; }
 
-int32_t AudioDeviceDummy::RecordingDelay(uint16_t& delayMS) const { return -1; }
-
 void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
 }  // namespace webrtc

modules/audio_device/dummy/audio_device_dummy.h

@@ -109,7 +109,6 @@ class AudioDeviceDummy : public AudioDeviceGeneric {
 
   // Delay information and control
   int32_t PlayoutDelay(uint16_t& delayMS) const override;
-  int32_t RecordingDelay(uint16_t& delayMS) const override;
 
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 };

modules/audio_device/dummy/file_audio_device.cc

@@ -395,8 +395,6 @@ int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
   return 0;
 }
 
-int32_t FileAudioDevice::RecordingDelay(uint16_t& delayMS) const { return -1; }
-
 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   rtc::CritScope lock(&_critSect);
 

modules/audio_device/dummy/file_audio_device.h

@@ -129,7 +129,6 @@ class FileAudioDevice : public AudioDeviceGeneric {
 
   // Delay information and control
   int32_t PlayoutDelay(uint16_t& delayMS) const override;
-  int32_t RecordingDelay(uint16_t& delayMS) const override;
 
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 

modules/audio_device/include/audio_device.h

@@ -19,6 +19,8 @@ namespace webrtc {
 
 class AudioDeviceModule : public rtc::RefCountInterface {
  public:
+  // Deprecated.
+  // TODO(henrika): to be removed.
   enum ErrorCode {
     kAdmErrNone = 0,
     kAdmErrArgument = 1
@@ -48,6 +50,7 @@ class AudioDeviceModule : public rtc::RefCountInterface {
 
  public:
   // Create an ADM.
+  // TODO(henrika): remove |id|.
   static rtc::scoped_refptr<AudioDeviceModule> Create(
       const int32_t id,
       const AudioLayer audio_layer);
@@ -55,8 +58,9 @@ class AudioDeviceModule : public rtc::RefCountInterface {
   // Retrieve the currently utilized audio layer
   virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
 
-  // Error handling
+  // Deprecated.
-  virtual ErrorCode LastError() const = 0;
+  // TODO(henrika): to be removed.
+  virtual ErrorCode LastError() const { return kAdmErrNone; }
 
   // Full-duplex transportation of PCM audio
   virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
@@ -142,9 +146,12 @@ class AudioDeviceModule : public rtc::RefCountInterface {
   virtual int32_t SetRecordingChannel(const ChannelType channel) = 0;
   virtual int32_t RecordingChannel(ChannelType* channel) const = 0;
 
-  // Delay information and control
+  // Playout delay
   virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
-  virtual int32_t RecordingDelay(uint16_t* delayMS) const = 0;
+
+  // Deprecated.
+  // TODO(henrika): to be removed.
+  virtual int32_t RecordingDelay(uint16_t* delayMS) const { return -1; }
 
   // Native sample rate controls (samples/sec)
   virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0;

modules/audio_device/include/fake_audio_device.h

@@ -42,7 +42,6 @@ class FakeAudioDeviceModule : public AudioDeviceModule {
   int32_t Terminate() override { return 0; }
 
   int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override { return 0; }
-  ErrorCode LastError() const override { return kAdmErrNone; }
   bool Initialized() const override { return true; }
   int16_t PlayoutDevices() override { return 0; }
   int16_t RecordingDevices() override { return 0; }
@@ -101,7 +100,6 @@ class FakeAudioDeviceModule : public AudioDeviceModule {
     *delayMS = 0;
     return 0;
   }
-  int32_t RecordingDelay(uint16_t* delayMS) const override { return 0; }
   int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override {
     return 0;
   }

modules/audio_device/include/mock_audio_device.h

@@ -86,7 +86,6 @@ class MockAudioDeviceModule : public AudioDeviceModule {
   MOCK_METHOD1(SetRecordingChannel, int32_t(const ChannelType channel));
   MOCK_CONST_METHOD1(RecordingChannel, int32_t(ChannelType* channel));
   MOCK_CONST_METHOD1(PlayoutDelay, int32_t(uint16_t* delayMS));
-  MOCK_CONST_METHOD1(RecordingDelay, int32_t(uint16_t* delayMS));
   MOCK_METHOD1(SetRecordingSampleRate, int32_t(const uint32_t samplesPerSec));
   MOCK_CONST_METHOD1(RecordingSampleRate, int32_t(uint32_t* samplesPerSec));
   MOCK_METHOD1(SetPlayoutSampleRate, int32_t(const uint32_t samplesPerSec));

modules/audio_device/ios/audio_device_ios.h

@@ -83,7 +83,6 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   // A/V-sync is not supported on iOS. However, we avoid adding error messages
   // the log by using these dummy implementations instead.
   int32_t PlayoutDelay(uint16_t& delayMS) const override;
-  int32_t RecordingDelay(uint16_t& delayMS) const override;
 
   // Native audio parameters stored during construction.
   // These methods are unique for the iOS implementation.

modules/audio_device/ios/audio_device_ios.mm

@@ -354,11 +354,6 @@ int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
   return 0;
 }
 
-int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const {
-  delayMS = kFixedRecordDelayEstimate;
-  return 0;
-}
-
 int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
   LOGI() << "GetPlayoutAudioParameters";
   RTC_DCHECK(playout_parameters_.is_valid());

modules/audio_device/linux/audio_device_alsa_linux.cc

@@ -1425,13 +1425,6 @@ int32_t AudioDeviceLinuxALSA::PlayoutDelay(uint16_t& delayMS) const
     return 0;
 }
 
-int32_t AudioDeviceLinuxALSA::RecordingDelay(uint16_t& delayMS) const
-{
-    // Adding 10ms adjusted value to the record delay due to 10ms buffering.
-    delayMS = (uint16_t)(10 + _recordingDelay * 1000 / _recordingFreq);
-    return 0;
-}
-
 bool AudioDeviceLinuxALSA::Playing() const
 {
     return (_playing);

modules/audio_device/linux/audio_device_alsa_linux.h

@@ -123,7 +123,6 @@ public:
 
     // Delay information and control
     int32_t PlayoutDelay(uint16_t& delayMS) const override;
-    int32_t RecordingDelay(uint16_t& delayMS) const override;
 
     void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 

modules/audio_device/linux/audio_device_pulse_linux.cc

@@ -1272,12 +1272,6 @@ int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const {
   return 0;
 }
 
-int32_t AudioDeviceLinuxPulse::RecordingDelay(uint16_t& delayMS) const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  delayMS = (uint16_t)_sndCardRecDelay;
-  return 0;
-}
-
 bool AudioDeviceLinuxPulse::Playing() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return (_playing);

modules/audio_device/linux/audio_device_pulse_linux.h

@@ -185,7 +185,6 @@ public:
 
     // Delay information and control
     int32_t PlayoutDelay(uint16_t& delayMS) const override;
-    int32_t RecordingDelay(uint16_t& delayMS) const override;
 
    void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 

modules/audio_device/mac/audio_device_mac.cc

@@ -1553,13 +1553,6 @@ int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const {
   return 0;
 }
 
-int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const {
-  int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-  delayMS =
-      static_cast<uint16_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
-  return 0;
-}
-
 bool AudioDeviceMac::Playing() const {
   return (_playing);
 }

modules/audio_device/mac/audio_device_mac.h

@@ -151,7 +151,6 @@ class AudioDeviceMac : public AudioDeviceGeneric {
 
   // Delay information and control
   virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
-  virtual int32_t RecordingDelay(uint16_t& delayMS) const;
 
   virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
 

modules/audio_device/win/audio_device_core_win.cc

@@ -3032,17 +3032,6 @@ int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
     return 0;
 }
 
-// ----------------------------------------------------------------------------
-//  RecordingDelay
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceWindowsCore::RecordingDelay(uint16_t& delayMS) const
-{
-    rtc::CritScope critScoped(&_critSect);
-    delayMS = static_cast<uint16_t>(_sndCardRecDelay);
-    return 0;
-}
-
 // ----------------------------------------------------------------------------
 //  Playing
 // ----------------------------------------------------------------------------

modules/audio_device/win/audio_device_core_win.h

@@ -173,7 +173,6 @@ public:
 
     // Delay information and control
     virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
-    virtual int32_t RecordingDelay(uint16_t& delayMS) const;
 
     virtual int32_t EnableBuiltInAEC(bool enable);
 

pc/test/fakeaudiocapturemodule.cc

@@ -74,11 +74,6 @@ int32_t FakeAudioCaptureModule::ActiveAudioLayer(
   return 0;
 }
 
-webrtc::AudioDeviceModule::ErrorCode FakeAudioCaptureModule::LastError() const {
-  RTC_NOTREACHED();
-  return webrtc::AudioDeviceModule::kAdmErrNone;
-}
-
 int32_t FakeAudioCaptureModule::RegisterAudioCallback(
     webrtc::AudioTransport* audio_callback) {
   rtc::CritScope cs(&crit_callback_);
@@ -417,11 +412,6 @@ int32_t FakeAudioCaptureModule::PlayoutDelay(uint16_t* delay_ms) const {
   return 0;
 }
 
-int32_t FakeAudioCaptureModule::RecordingDelay(uint16_t* /*delay_ms*/) const {
-  RTC_NOTREACHED();
-  return 0;
-}
-
 int32_t FakeAudioCaptureModule::SetRecordingSampleRate(
     const uint32_t /*samples_per_sec*/) {
   RTC_NOTREACHED();

pc/test/fakeaudiocapturemodule.h

@@ -54,8 +54,6 @@ class FakeAudioCaptureModule
 
   int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
 
-  ErrorCode LastError() const override;
-
   // Note: Calling this method from a callback may result in deadlock.
   int32_t RegisterAudioCallback(
       webrtc::AudioTransport* audio_callback) override;
@@ -131,7 +129,6 @@ class FakeAudioCaptureModule
   int32_t RecordingChannel(ChannelType* channel) const override;
 
   int32_t PlayoutDelay(uint16_t* delay_ms) const override;
-  int32_t RecordingDelay(uint16_t* delay_ms) const override;
 
   int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
   int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;