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}
2017-10-17 14:47:44 +02:00 · 2017-10-17 14:47:44 +02:00 · 6592f2cfd2
commit 6592f2cfd2
parent 0743814fb8
25 changed files with 13 additions and 108 deletions
--- a/media/engine/webrtcvoiceengine.cc
+++ b/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="
-                      << adm()->LastError();
+                      << *options.recording_sample_rate << ") failed.";
    }
  }

@ -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="
-                      << adm()->LastError();
+                      << *options.playout_sample_rate << ") failed.";
    }
  }
  return true;
--- a/modules/audio_device/android/audio_device_template.h
+++ b/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);
--- a/modules/audio_device/audio_device_data_observer.cc
+++ b/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);
  }
--- a/modules/audio_device/audio_device_generic.h
+++ b/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);
--- a/modules/audio_device/audio_device_impl.cc
+++ b/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 << ")";
--- a/modules/audio_device/audio_device_impl.h
+++ b/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;
--- a/modules/audio_device/dummy/audio_device_dummy.cc
+++ b/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
--- a/modules/audio_device/dummy/audio_device_dummy.h
+++ b/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;
 };
--- a/modules/audio_device/dummy/file_audio_device.cc
+++ b/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);

--- a/modules/audio_device/dummy/file_audio_device.h
+++ b/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;

--- a/modules/audio_device/include/audio_device.h
+++ b/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
-  virtual ErrorCode LastError() const = 0;
+  // Deprecated.
+  // 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;
--- a/modules/audio_device/include/fake_audio_device.h
+++ b/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;
  }
--- a/modules/audio_device/include/mock_audio_device.h
+++ b/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));
--- a/modules/audio_device/ios/audio_device_ios.h
+++ b/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.
--- a/modules/audio_device/ios/audio_device_ios.mm
+++ b/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());
--- a/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/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);
--- a/modules/audio_device/linux/audio_device_alsa_linux.h
+++ b/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;

--- a/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/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);
--- a/modules/audio_device/linux/audio_device_pulse_linux.h
+++ b/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;

--- a/modules/audio_device/mac/audio_device_mac.cc
+++ b/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);
 }
--- a/modules/audio_device/mac/audio_device_mac.h
+++ b/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);

--- a/modules/audio_device/win/audio_device_core_win.cc
+++ b/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
 // ----------------------------------------------------------------------------
--- a/modules/audio_device/win/audio_device_core_win.h
+++ b/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);

--- a/pc/test/fakeaudiocapturemodule.cc
+++ b/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();
--- a/pc/test/fakeaudiocapturemodule.h
+++ b/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;