Refactoring for typing detection

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_device/audio_device_buffer.cc

@@ -46,6 +46,7 @@ AudioDeviceBuffer::AudioDeviceBuffer() :
     _playFile(*FileWrapper::Create()),
     _currentMicLevel(0),
     _newMicLevel(0),
+    _typingStatus(false),
     _playDelayMS(0),
     _recDelayMS(0),
     _clockDrift(0) {
@@ -266,6 +267,12 @@ int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level)
     return 0;
 }
 
+int32_t AudioDeviceBuffer::SetTypingStatus(bool typingStatus)
+{
+    _typingStatus = typingStatus;
+    return 0;
+}
+
 // ----------------------------------------------------------------------------
 //  NewMicLevel
 // ----------------------------------------------------------------------------
@@ -469,6 +476,7 @@ int32_t AudioDeviceBuffer::DeliverRecordedData()
                                                         totalDelayMS,
                                                         _clockDrift,
                                                         _currentMicLevel,
+                                                        _typingStatus,
                                                         newMicLevel);
     if (res != -1)
     {

webrtc/modules/audio_device/audio_device_buffer.h

@@ -67,6 +67,8 @@ public:
         const char fileName[kAdmMaxFileNameSize]);
     int32_t StopOutputFileRecording();
 
+    int32_t SetTypingStatus(bool typingStatus);
+
     AudioDeviceBuffer();
     ~AudioDeviceBuffer();
 
@@ -110,6 +112,8 @@ private:
     uint32_t                  _currentMicLevel;
     uint32_t                  _newMicLevel;
 
+    bool                      _typingStatus;
+
     uint32_t                  _playDelayMS;
     uint32_t                  _recDelayMS;
 

webrtc/modules/audio_device/include/audio_device_defines.h

@@ -62,6 +62,7 @@ public:
                                             const uint32_t totalDelayMS,
                                             const int32_t clockDrift,
                                             const uint32_t currentMicLevel,
+                                            const bool keyPressed,
                                             uint32_t& newMicLevel) = 0;   
 
     virtual int32_t NeedMorePlayData(const uint32_t nSamples,

webrtc/modules/audio_device/mac/audio_device_mac.cc

@@ -13,15 +13,17 @@
 #include "audio_device_config.h"
 
 #include "event_wrapper.h"
+#include "portaudio/pa_ringbuffer.h"
 #include "trace.h"
 #include "thread_wrapper.h"
 
+#include <ApplicationServices/ApplicationServices.h>
 #include <cassert>
-
-#include <sys/sysctl.h>         // sysctlbyname()
-#include <mach/mach.h>          // mach_task_self()
 #include <libkern/OSAtomic.h>   // OSAtomicCompareAndSwap()
-#include "portaudio/pa_ringbuffer.h"
+#include <mach/mach.h>          // mach_task_self()
+#include <sys/sysctl.h>         // sysctlbyname()
+
+
 
 namespace webrtc
 {
@@ -3207,6 +3209,8 @@ bool AudioDeviceMac::CaptureWorkerThread()
 
         _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
 
+        _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
         // deliver recorded samples at specified sample rate, mic level etc.
         // to the observer using callback
         _ptrAudioBuffer->DeliverRecordedData();
@@ -3236,4 +3240,14 @@ bool AudioDeviceMac::CaptureWorkerThread()
     return true;
 }
 
+bool AudioDeviceMac::KeyPressed() const{
+
+  bool key_down = false;
+  // loop through all Mac virtual key constant values
+  for (int key_index = 0; key_index <= 0x5C; key_index++) {
+    key_down |= CGEventSourceKeyState(kCGEventSourceStateHIDSystemState,
+                                      key_index);
+  }
+  return(key_down);
+}
 } //  namespace webrtc

webrtc/modules/audio_device/mac/audio_device_mac.h

@@ -293,6 +293,9 @@ private:
     bool CaptureWorkerThread();
     bool RenderWorkerThread();
 
+private:
+    bool KeyPressed() const;
+
 private:
     AudioDeviceBuffer* _ptrAudioBuffer;
 

webrtc/modules/audio_device/test/audio_device_test_api.cc

@@ -92,6 +92,7 @@ class AudioTransportAPI: public AudioTransport {
       const uint32_t totalDelay,
       const int32_t clockSkew,
       const uint32_t currentMicLevel,
+      const bool keyPressed,
       uint32_t& newMicLevel) {
     rec_count_++;
     if (rec_count_ % 100 == 0) {

webrtc/modules/audio_device/test/func_test_manager.cc

@@ -176,6 +176,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
     const uint32_t totalDelayMS,
     const int32_t clockDrift,
     const uint32_t currentMicLevel,
+    const bool keyPressed,
     uint32_t& newMicLevel)
 {
     if (_fullDuplex && _audioList.GetSize() < 15)

webrtc/modules/audio_device/test/func_test_manager.h

@@ -101,6 +101,7 @@ public:
                                 const uint32_t totalDelayMS,
                                 const int32_t clockDrift,
                                 const uint32_t currentMicLevel,
+                                const bool keyPressed,
                                 uint32_t& newMicLevel);
 
     virtual int32_t NeedMorePlayData(const uint32_t nSamples,

webrtc/modules/audio_device/win/audio_device_core_win.cc

@@ -4092,6 +4092,8 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
                                                     sndCardRecDelay,
                                                     0);
 
+                        _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
                         QueryPerformanceCounter(&t1);    // measure time: START
 
                         _UnLock();  // release lock while making the callback
@@ -5145,6 +5147,16 @@ char* AudioDeviceWindowsCore::WideToUTF8(const TCHAR* src) const {
 #endif
 }
 
+
+bool AudioDeviceWindowsCore::KeyPressed() const{
+
+  int key_down = 0;
+  for (int key = VK_SPACE; key < VK_NUMLOCK; key++) {
+    short res = GetAsyncKeyState(key);
+    key_down |= res & 0x1; // Get the LSB
+  }
+  return (key_down > 0);
+}
 }  // namespace webrtc
 
 #endif  // WEBRTC_WINDOWS_CORE_AUDIO_BUILD

webrtc/modules/audio_device/win/audio_device_core_win.h

@@ -210,6 +210,9 @@ public:
 public:
     virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
 
+private:
+    bool KeyPressed() const;
+
 private:    // avrt function pointers
     PAvRevertMmThreadCharacteristics    _PAvRevertMmThreadCharacteristics;
     PAvSetMmThreadCharacteristicsA      _PAvSetMmThreadCharacteristicsA;

webrtc/modules/audio_device/win/audio_device_wave_win.cc

@@ -3359,6 +3359,8 @@ int32_t AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
 
         _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, drift);
 
+        _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
         // Store the play and rec delay values for video synchronization
         _sndCardPlayDelay = msecOnPlaySide;
         _sndCardRecDelay = msecOnRecordSide;
@@ -3820,5 +3822,15 @@ int32_t AudioDeviceWindowsWave::RestartTimerIfNeeded(const uint32_t time)
     return 0;
 }
 
+
+bool AudioDeviceWindowsWave::KeyPressed() const{
+
+  int key_down = 0;
+  for (int key = VK_SPACE; key < VK_NUMLOCK; key++) {
+    short res = GetAsyncKeyState(key);
+    key_down |= res & 0x1; // Get the LSB
+  }
+  return (key_down > 0);
+}
 }  // namespace webrtc
 

webrtc/modules/audio_device/win/audio_device_wave_win.h

@@ -177,6 +177,9 @@ private:
     inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
     inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
 
+private:
+    bool KeyPressed() const;
+
 private:
     int32_t EnumeratePlayoutDevices();
     int32_t EnumerateRecordingDevices();

webrtc/system_wrappers/interface/event_wrapper.h

@@ -54,12 +54,6 @@ class EventWrapper {
 
   virtual bool StopTimer() = 0;
 
-  // Only implemented on Windows
-  // Returns 1 if a key has been pressed since last call to this function.
-  // -1 indicates failure
-  // 0 indicates no key has been pressed since last call
-  // TODO(hellner) this function does not seem to belong here
-  static int KeyPressed();
 };
 } // namespace webrtc
 

webrtc/system_wrappers/source/event.cc

@@ -30,33 +30,4 @@ EventWrapper* EventWrapper::Create() {
   return EventPosix::Create();
 #endif
 }
-
-int EventWrapper::KeyPressed() {
-#if defined(_WIN32)
-  int key_down = 0;
-  for (int key = 0x20; key < 0x90; key++) {
-    short res = GetAsyncKeyState(key);
-    key_down |= res % 2; // Get the LSB
-  }
-  if (key_down) {
-    return 1;
-  } else {
-    return 0;
-  }
-#elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-  bool key_down = false;
-  // loop through all Mac virtual key constant values
-  for (int key_index = 0; key_index <= 0x5C; key_index++) {
-    key_down |= CGEventSourceKeyState(kCGEventSourceStateHIDSystemState,
-                                      key_index);
-  }
-  if (key_down) {
-    return 1;
-  } else {
-    return 0;
-  }
-#else
-  return -1;
-#endif
-}
 } // namespace webrtc

webrtc/voice_engine/transmit_mixer.cc

@@ -332,7 +332,8 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
                             const uint32_t samplesPerSec,
                             const uint16_t totalDelayMS,
                             const int32_t clockDrift,
-                            const uint16_t currentMicLevel)
+                            const uint16_t currentMicLevel,
+                            const bool keyPressed)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
@@ -369,7 +370,7 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
 
     // --- Annoying typing detection (utilizes the APM/VAD decision)
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    TypingDetection();
+    TypingDetection(keyPressed);
 #endif
 
     // --- Mute during DTMF tone if direct feedback is enabled
@@ -1327,7 +1328,7 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-int TransmitMixer::TypingDetection()
+int TransmitMixer::TypingDetection(const bool keyPressed)
 {
 
     // We let the VAD determine if we're using this feature or not.
@@ -1336,13 +1337,6 @@ int TransmitMixer::TypingDetection()
         return (0);
     }
 
-    int keyPressed = EventWrapper::KeyPressed();
-
-    if (keyPressed < 0)
-    {
-        return (-1);
-    }
-
     if (_audioFrame.vad_activity_ == AudioFrame::kVadActive)
         _timeActive++;
     else

webrtc/voice_engine/transmit_mixer.h

@@ -56,7 +56,8 @@ public:
                          const uint32_t samplesPerSec,
                          const uint16_t totalDelayMS,
                          const int32_t  clockDrift,
-                         const uint16_t currentMicLevel);
+                         const uint16_t currentMicLevel,
+                         const bool keyPressed);
 
 
     int32_t DemuxAndMix();
@@ -178,7 +179,7 @@ private:
     void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level);
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    int TypingDetection();
+    int TypingDetection(const bool keyPressed);
 #endif
 
     // uses

webrtc/voice_engine/voe_base_impl.cc

@@ -133,6 +133,7 @@ int32_t VoEBaseImpl::RecordedDataIsAvailable(
         const uint32_t totalDelayMS,
         const int32_t clockDrift,
         const uint32_t currentMicLevel,
+        const bool keyPressed,
         uint32_t& newMicLevel)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
@@ -195,7 +196,7 @@ int32_t VoEBaseImpl::RecordedDataIsAvailable(
     // (APM, mix with file, record to file, mute, etc.)
     _shared->transmit_mixer()->PrepareDemux(audioSamples, nSamples, nChannels,
         samplesPerSec, static_cast<uint16_t>(totalDelayMS), clockDrift,
-        currentVoEMicLevel);
+        currentVoEMicLevel, keyPressed);
 
     // Copy the audio frame to each sending channel and perform
     // channel-dependent operations (file mixing, mute, etc.) to prepare

webrtc/voice_engine/voe_base_impl.h

@@ -80,6 +80,7 @@ public:
                                 const uint32_t totalDelayMS,
                                 const int32_t clockDrift,
                                 const uint32_t currentMicLevel,
+                                const bool keyPressed,
                                 uint32_t& newMicLevel);
 
     virtual int32_t NeedMorePlayData(const uint32_t nSamples,

webrtc/voice_engine/voe_external_media_impl.cc

@@ -244,7 +244,8 @@ int VoEExternalMediaImpl::ExternalRecordingInsertData(
             samplingFreqHz,
             totalDelayMS,
             0,
-            0);
+            0,
+            false); // Typing detection not supported
 
         shared_->transmit_mixer()->DemuxAndMix();
         shared_->transmit_mixer()->EncodeAndSend();