Small refactoring of AudioProcessing use in channel.cc.

- Apply consistent naming. - Use a scoped_ptr for rx_audioproc_. - Remove now unnecessary AudioProcessing::Destroy(). R=bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2184007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4784 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-18 22:37:32 +00:00 · 2013-09-18 22:37:32 +00:00 · f3930e941c
commit f3930e941c
parent 0d5da25e6c
6 changed files with 81 additions and 135 deletions
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@ -47,10 +47,6 @@ AudioProcessing* AudioProcessing::Create(int id) {
  return apm;
 }

-void AudioProcessing::Destroy(AudioProcessing* apm) {
-  delete static_cast<AudioProcessingImpl*>(apm);
-}
-
 int32_t AudioProcessing::TimeUntilNextProcess() { return -1; }
 int32_t AudioProcessing::Process() { return -1; }

--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@ -105,8 +105,7 @@ class VoiceDetection;
 // apm->Initialize();
 //
 // // Close the application...
-// AudioProcessing::Destroy(apm);
-// apm = NULL;
+// delete apm;
 //
 class AudioProcessing : public Module {
 public:
@ -118,11 +117,6 @@ class AudioProcessing : public Module {
  static AudioProcessing* Create(int id);
  virtual ~AudioProcessing() {}

-  // TODO(andrew): remove this method. We now allow users to delete instances
-  // directly, useful for scoped_ptr.
-  // Destroys a |apm| instance.
-  static void Destroy(AudioProcessing* apm);
-
  // Initializes internal states, while retaining all user settings. This
  // should be called before beginning to process a new audio stream. However,
  // it is not necessary to call before processing the first stream after
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@ -42,6 +42,7 @@ using webrtc::NoiseSuppression;
 using webrtc::EchoCancellation;
 using webrtc::EventWrapper;
 using webrtc::scoped_array;
+using webrtc::scoped_ptr;
 using webrtc::Trace;
 using webrtc::LevelEstimator;
 using webrtc::EchoCancellation;
@ -244,9 +245,9 @@ class ApmTest : public ::testing::Test {
  const std::string output_path_;
  const std::string ref_path_;
  const std::string ref_filename_;
-  webrtc::AudioProcessing* apm_;
-  webrtc::AudioFrame* frame_;
-  webrtc::AudioFrame* revframe_;
+  scoped_ptr<webrtc::AudioProcessing> apm_;
+  AudioFrame* frame_;
+  AudioFrame* revframe_;
  FILE* far_file_;
  FILE* near_file_;
  FILE* out_file_;
@ -261,7 +262,7 @@ ApmTest::ApmTest()
 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
      ref_filename_(ref_path_ + "output_data_float.pb"),
 #endif
-      apm_(NULL),
+      apm_(AudioProcessing::Create(0)),
      frame_(NULL),
      revframe_(NULL),
      far_file_(NULL),
@ -269,8 +270,7 @@ ApmTest::ApmTest()
      out_file_(NULL) {}

 void ApmTest::SetUp() {
-  apm_ = AudioProcessing::Create(0);
-  ASSERT_TRUE(apm_ != NULL);
+  ASSERT_TRUE(apm_.get() != NULL);

  frame_ = new AudioFrame();
  revframe_ = new AudioFrame();
@ -303,11 +303,6 @@ void ApmTest::TearDown() {
    ASSERT_EQ(0, fclose(out_file_));
  }
  out_file_ = NULL;
-
-  if (apm_ != NULL) {
-    AudioProcessing::Destroy(apm_);
-  }
-  apm_ = NULL;
 }

 std::string ApmTest::ResourceFilePath(std::string name, int sample_rate_hz) {
@ -466,8 +461,8 @@ void ApmTest::ChangeTriggersInit(F f, AudioProcessing* ap, int initial_value,
  EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));

  // Test that a change in value triggers an init.
-  f(apm_, changed_value);
-  f(apm_, initial_value);
+  f(apm_.get(), changed_value);
+  f(apm_.get(), initial_value);
  ProcessWithDefaultStreamParameters(&frame_copy);
  EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));

@ -492,7 +487,7 @@ void ApmTest::ChangeTriggersInit(F f, AudioProcessing* ap, int initial_value,
  apm_->Initialize();
  ProcessWithDefaultStreamParameters(&frame_copy);
  // Test that the same value does not trigger an init.
-  f(apm_, initial_value);
+  f(apm_.get(), initial_value);
  ProcessWithDefaultStreamParameters(&frame_copy);
  EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
 }
@ -737,15 +732,15 @@ void SetNumOutputChannels(AudioProcessing* ap, int value) {
 }

 TEST_F(ApmTest, SampleRateChangeTriggersInit) {
-  ChangeTriggersInit(SetSampleRate, apm_, 16000, 8000);
+  ChangeTriggersInit(SetSampleRate, apm_.get(), 16000, 8000);
 }

 TEST_F(ApmTest, ReverseChannelChangeTriggersInit) {
-  ChangeTriggersInit(SetNumReverseChannels, apm_, 2, 1);
+  ChangeTriggersInit(SetNumReverseChannels, apm_.get(), 2, 1);
 }

 TEST_F(ApmTest, ChannelChangeTriggersInit) {
-  ChangeTriggersInit(SetNumOutputChannels, apm_, 2, 1);
+  ChangeTriggersInit(SetNumOutputChannels, apm_.get(), 2, 1);
 }

 TEST_F(ApmTest, EchoCancellation) {
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@ -38,6 +38,7 @@ using webrtc::EchoCancellation;
 using webrtc::GainControl;
 using webrtc::NoiseSuppression;
 using webrtc::scoped_array;
+using webrtc::scoped_ptr;
 using webrtc::TickInterval;
 using webrtc::TickTime;
 using webrtc::VoiceDetection;
@ -167,8 +168,8 @@ void void_main(int argc, char* argv[]) {
    printf("Try `process_test --help' for more information.\n\n");
  }

-  AudioProcessing* apm = AudioProcessing::Create(0);
-  ASSERT_TRUE(apm != NULL);
+  scoped_ptr<AudioProcessing> apm(AudioProcessing::Create(0));
+  ASSERT_TRUE(apm.get() != NULL);

  const char* pb_filename = NULL;
  const char* far_filename = NULL;
@ -1057,9 +1058,6 @@ void void_main(int argc, char* argv[]) {
      printf("Warning: no capture frames\n");
    }
  }
-
-  AudioProcessing::Destroy(apm);
-  apm = NULL;
 }
 }  // namespace

--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@ -53,11 +53,10 @@ Channel::SendData(FrameType frameType,

    if (_includeAudioLevelIndication)
    {
-        assert(_rtpAudioProc.get() != NULL);
        // Store current audio level in the RTP/RTCP module.
        // The level will be used in combination with voice-activity state
        // (frameType) to add an RTP header extension
-        _rtpRtcpModule->SetAudioLevel(_rtpAudioProc->level_estimator()->RMS());
+        _rtpRtcpModule->SetAudioLevel(rtp_audioproc_->level_estimator()->RMS());
    }

    // Push data from ACM to RTP/RTCP-module to deliver audio frame for
@ -960,8 +959,8 @@ Channel::Channel(int32_t channelId,
    _callbackCritSectPtr(NULL),
    _transportPtr(NULL),
    _encryptionPtr(NULL),
-    _rtpAudioProc(NULL),
-    _rxAudioProcessingModulePtr(NULL),
+    rtp_audioproc_(NULL),
+    rx_audioproc_(AudioProcessing::Create(VoEModuleId(instanceId, channelId))),
    _rxVadObserverPtr(NULL),
    _oldVadDecision(-1),
    _sendFrameType(0),
@ -1025,10 +1024,6 @@ Channel::Channel(int32_t channelId,
    configuration.receive_statistics = rtp_receive_statistics_.get();

    _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
-
-    // Create far end AudioProcessing Module
-    _rxAudioProcessingModulePtr = AudioProcessing::Create(
-        VoEModuleId(instanceId, channelId));
 }

 Channel::~Channel()
@ -1100,11 +1095,6 @@ Channel::~Channel()

    // Destroy modules
    AudioCodingModule::Destroy(&_audioCodingModule);
-    if (_rxAudioProcessingModulePtr != NULL)
-    {
-        AudioProcessing::Destroy(_rxAudioProcessingModulePtr); // far end APM
-        _rxAudioProcessingModulePtr = NULL;
-    }

    // End of modules shutdown

@ -1277,19 +1267,8 @@ Channel::Init()
 #endif
    }

-    // Initialize the far end AP module
-    // Using 8 kHz as initial Fs, the same as in transmission. Might be
-    // changed at the first receiving audio.
-    if (_rxAudioProcessingModulePtr == NULL)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_NO_MEMORY, kTraceCritical,
-            "Channel::Init() failed to create the far-end AudioProcessing"
-            " module");
-        return -1;
-    }

-    if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000))
+    if (rx_audioproc_->set_sample_rate_hz(8000))
    {
        _engineStatisticsPtr->SetLastError(
            VE_APM_ERROR, kTraceWarning,
@ -1297,14 +1276,14 @@ Channel::Init()
            " far-end AP module");
    }

-    if (_rxAudioProcessingModulePtr->set_num_channels(1, 1) != 0)
+    if (rx_audioproc_->set_num_channels(1, 1) != 0)
    {
        _engineStatisticsPtr->SetLastError(
            VE_SOUNDCARD_ERROR, kTraceWarning,
            "Init() failed to set channels for the primary audio stream");
    }

-    if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable(
+    if (rx_audioproc_->high_pass_filter()->Enable(
        WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -1313,7 +1292,7 @@ Channel::Init()
            " far-end AP module");
    }

-    if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(
+    if (rx_audioproc_->noise_suppression()->set_level(
        (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -1321,7 +1300,7 @@ Channel::Init()
            "Init() failed to set noise reduction level for far-end"
            " AP module");
    }
-    if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(
+    if (rx_audioproc_->noise_suppression()->Enable(
        WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -1330,14 +1309,14 @@ Channel::Init()
            " AP module");
    }

-    if (_rxAudioProcessingModulePtr->gain_control()->set_mode(
+    if (rx_audioproc_->gain_control()->set_mode(
        (GainControl::Mode)WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE) != 0)
    {
        _engineStatisticsPtr->SetLastError(
            VE_APM_ERROR, kTraceWarning,
            "Init() failed to set AGC mode for far-end AP module");
    }
-    if (_rxAudioProcessingModulePtr->gain_control()->Enable(
+    if (rx_audioproc_->gain_control()->Enable(
        WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -3317,7 +3296,7 @@ Channel::SetRxAgcStatus(bool enable, AgcModes mode)
            agcMode = GainControl::kAdaptiveDigital;
            break;
        case kAgcUnchanged:
-            agcMode = _rxAudioProcessingModulePtr->gain_control()->mode();
+            agcMode = rx_audioproc_->gain_control()->mode();
            break;
        case kAgcFixedDigital:
            agcMode = GainControl::kFixedDigital;
@ -3332,14 +3311,14 @@ Channel::SetRxAgcStatus(bool enable, AgcModes mode)
            return -1;
    }

-    if (_rxAudioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0)
+    if (rx_audioproc_->gain_control()->set_mode(agcMode) != 0)
    {
        _engineStatisticsPtr->SetLastError(
            VE_APM_ERROR, kTraceError,
            "SetRxAgcStatus() failed to set Agc mode");
        return -1;
    }
-    if (_rxAudioProcessingModulePtr->gain_control()->Enable(enable) != 0)
+    if (rx_audioproc_->gain_control()->Enable(enable) != 0)
    {
        _engineStatisticsPtr->SetLastError(
            VE_APM_ERROR, kTraceError,
@ -3359,9 +3338,9 @@ Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                     "Channel::GetRxAgcStatus(enable=?, mode=?)");

-    bool enable = _rxAudioProcessingModulePtr->gain_control()->is_enabled();
+    bool enable = rx_audioproc_->gain_control()->is_enabled();
    GainControl::Mode agcMode =
-        _rxAudioProcessingModulePtr->gain_control()->mode();
+        rx_audioproc_->gain_control()->mode();

    enabled = enable;

@ -3389,7 +3368,7 @@ Channel::SetRxAgcConfig(AgcConfig config)
    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                 "Channel::SetRxAgcConfig()");

-    if (_rxAudioProcessingModulePtr->gain_control()->set_target_level_dbfs(
+    if (rx_audioproc_->gain_control()->set_target_level_dbfs(
        config.targetLeveldBOv) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -3398,7 +3377,7 @@ Channel::SetRxAgcConfig(AgcConfig config)
            "(or envelope) of the Agc");
        return -1;
    }
-    if (_rxAudioProcessingModulePtr->gain_control()->set_compression_gain_db(
+    if (rx_audioproc_->gain_control()->set_compression_gain_db(
        config.digitalCompressionGaindB) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -3407,7 +3386,7 @@ Channel::SetRxAgcConfig(AgcConfig config)
            " digital compression stage may apply");
        return -1;
    }
-    if (_rxAudioProcessingModulePtr->gain_control()->enable_limiter(
+    if (rx_audioproc_->gain_control()->enable_limiter(
        config.limiterEnable) != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -3426,11 +3405,11 @@ Channel::GetRxAgcConfig(AgcConfig& config)
                 "Channel::GetRxAgcConfig(config=%?)");

    config.targetLeveldBOv =
-        _rxAudioProcessingModulePtr->gain_control()->target_level_dbfs();
+        rx_audioproc_->gain_control()->target_level_dbfs();
    config.digitalCompressionGaindB =
-        _rxAudioProcessingModulePtr->gain_control()->compression_gain_db();
+        rx_audioproc_->gain_control()->compression_gain_db();
    config.limiterEnable =
-        _rxAudioProcessingModulePtr->gain_control()->is_limiter_enabled();
+        rx_audioproc_->gain_control()->is_limiter_enabled();

    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
               VoEId(_instanceId,_channelId), "GetRxAgcConfig() => "
@ -3464,7 +3443,7 @@ Channel::SetRxNsStatus(bool enable, NsModes mode)
                WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE;
            break;
        case kNsUnchanged:
-            nsLevel = _rxAudioProcessingModulePtr->noise_suppression()->level();
+            nsLevel = rx_audioproc_->noise_suppression()->level();
            break;
        case kNsConference:
            nsLevel = NoiseSuppression::kHigh;
@ -3483,7 +3462,7 @@ Channel::SetRxNsStatus(bool enable, NsModes mode)
            break;
    }

-    if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(nsLevel)
+    if (rx_audioproc_->noise_suppression()->set_level(nsLevel)
        != 0)
    {
        _engineStatisticsPtr->SetLastError(
@ -3491,7 +3470,7 @@ Channel::SetRxNsStatus(bool enable, NsModes mode)
            "SetRxAgcStatus() failed to set Ns level");
        return -1;
    }
-    if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(enable) != 0)
+    if (rx_audioproc_->noise_suppression()->Enable(enable) != 0)
    {
        _engineStatisticsPtr->SetLastError(
            VE_APM_ERROR, kTraceError,
@ -3512,9 +3491,9 @@ Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
                 "Channel::GetRxNsStatus(enable=?, mode=?)");

    bool enable =
-        _rxAudioProcessingModulePtr->noise_suppression()->is_enabled();
+        rx_audioproc_->noise_suppression()->is_enabled();
    NoiseSuppression::Level ncLevel =
-        _rxAudioProcessingModulePtr->noise_suppression()->level();
+        rx_audioproc_->noise_suppression()->level();

    enabled = enable;

@ -3702,34 +3681,28 @@ Channel::GetRemoteCSRCs(unsigned int arrCSRC[15])
 int
 Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID)
 {
-    if (_rtpAudioProc.get() == NULL)
-    {
-        _rtpAudioProc.reset(AudioProcessing::Create(VoEModuleId(_instanceId,
-                                                                _channelId)));
-        if (_rtpAudioProc.get() == NULL)
-        {
-            _engineStatisticsPtr->SetLastError(VE_NO_MEMORY, kTraceCritical,
-                "Failed to create AudioProcessing");
-            return -1;
-        }
-    }
+  if (rtp_audioproc_.get() == NULL) {
+    rtp_audioproc_.reset(AudioProcessing::Create(VoEModuleId(_instanceId,
+                                                             _channelId)));
+  }

-    if (_rtpAudioProc->level_estimator()->Enable(enable) !=
-        AudioProcessing::kNoError)
-    {
-        _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceWarning,
-            "Failed to enable AudioProcessing::level_estimator()");
-    }
+  if (rtp_audioproc_->level_estimator()->Enable(enable) !=
+      AudioProcessing::kNoError) {
+    _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceError,
+        "Failed to enable AudioProcessing::level_estimator()");
+    return -1;
+  }

-    _includeAudioLevelIndication = enable;
-    if (enable) {
-      rtp_header_parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
-                                                     ID);
-    } else {
-      rtp_header_parser_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
-    }
-    return _rtpRtcpModule->SetRTPAudioLevelIndicationStatus(enable, ID);
+  _includeAudioLevelIndication = enable;
+  if (enable) {
+    rtp_header_parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                   ID);
+  } else {
+    rtp_header_parser_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
+  }
+  return _rtpRtcpModule->SetRTPAudioLevelIndicationStatus(enable, ID);
 }
+
 int
 Channel::GetRTPAudioLevelIndicationStatus(bool& enabled, unsigned char& ID)
 {
@ -4506,36 +4479,27 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)

    if (_includeAudioLevelIndication)
    {
-        assert(_rtpAudioProc.get() != NULL);
-
-        // Check if settings need to be updated.
-        if (_rtpAudioProc->sample_rate_hz() != _audioFrame.sample_rate_hz_)
+        if (rtp_audioproc_->set_sample_rate_hz(_audioFrame.sample_rate_hz_) !=
+            AudioProcessing::kNoError)
        {
-            if (_rtpAudioProc->set_sample_rate_hz(_audioFrame.sample_rate_hz_) !=
-                AudioProcessing::kNoError)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                             VoEId(_instanceId, _channelId),
-                             "Error setting AudioProcessing sample rate");
-                return -1;
-            }
+            WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+                         VoEId(_instanceId, _channelId),
+                         "Error setting AudioProcessing sample rate");
+            return -1;
        }

-        if (_rtpAudioProc->num_input_channels() != _audioFrame.num_channels_)
+        if (rtp_audioproc_->set_num_channels(_audioFrame.num_channels_,
+                                             _audioFrame.num_channels_) !=
+            AudioProcessing::kNoError)
        {
-            if (_rtpAudioProc->set_num_channels(_audioFrame.num_channels_,
-                                                _audioFrame.num_channels_)
-                != AudioProcessing::kNoError)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                             VoEId(_instanceId, _channelId),
-                             "Error setting AudioProcessing channels");
-                return -1;
-            }
+            WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+                         VoEId(_instanceId, _channelId),
+                         "Error setting AudioProcessing channels");
+            return -1;
        }

        // Performs level analysis only; does not affect the signal.
-        _rtpAudioProc->ProcessStream(&_audioFrame);
+        rtp_audioproc_->ProcessStream(&_audioFrame);
    }

    return 0;
@ -5271,16 +5235,15 @@ Channel::RegisterReceiveCodecsToRTPModule()
 }

 int Channel::ApmProcessRx(AudioFrame& frame) {
-  AudioProcessing* audioproc = _rxAudioProcessingModulePtr;
  // Register the (possibly new) frame parameters.
-  if (audioproc->set_sample_rate_hz(frame.sample_rate_hz_) != 0) {
+  if (rx_audioproc_->set_sample_rate_hz(frame.sample_rate_hz_) != 0) {
    LOG_FERR1(LS_WARNING, set_sample_rate_hz, frame.sample_rate_hz_);
  }
-  if (audioproc->set_num_channels(frame.num_channels_,
-                                  frame.num_channels_) != 0) {
+  if (rx_audioproc_->set_num_channels(frame.num_channels_,
+                                      frame.num_channels_) != 0) {
    LOG_FERR1(LS_WARNING, set_num_channels, frame.num_channels_);
  }
-  if (audioproc->ProcessStream(&frame) != 0) {
+  if (rx_audioproc_->ProcessStream(&frame) != 0) {
    LOG_FERR0(LS_WARNING, ProcessStream);
  }
  return 0;
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@ -520,8 +520,8 @@ private:
    CriticalSectionWrapper* _callbackCritSectPtr; // owned by base
    Transport* _transportPtr; // WebRtc socket or external transport
    Encryption* _encryptionPtr; // WebRtc SRTP or external encryption
-    scoped_ptr<AudioProcessing> _rtpAudioProc;
-    AudioProcessing* _rxAudioProcessingModulePtr; // far end AudioProcessing
+    scoped_ptr<AudioProcessing> rtp_audioproc_;
+    scoped_ptr<AudioProcessing> rx_audioproc_; // far end AudioProcessing
    VoERxVadCallback* _rxVadObserverPtr;
    int32_t _oldVadDecision;
    int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise