generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Refactoring for typing detection

Browse Source 
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
This commit is contained in:
niklas.enbom@webrtc.org 2013-05-07 21:04:24 +00:00
parent ef14488d03
commit 3be565b502
19 changed files with 79 additions and 53 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_device/audio_device_buffer.cc
View File

@ -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)
{

4
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -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;

1
webrtc/modules/audio_device/include/audio_device_defines.h
View File

@ -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,

22
webrtc/modules/audio_device/mac/audio_device_mac.cc
View File

@ -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

3
webrtc/modules/audio_device/mac/audio_device_mac.h
View File

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

1
webrtc/modules/audio_device/test/audio_device_test_api.cc
View File

@ -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) {

1
webrtc/modules/audio_device/test/func_test_manager.cc
View File

@ -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)

1
webrtc/modules/audio_device/test/func_test_manager.h
View File

@ -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,

12
webrtc/modules/audio_device/win/audio_device_core_win.cc
View File

@ -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

3
webrtc/modules/audio_device/win/audio_device_core_win.h
View File

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

12
webrtc/modules/audio_device/win/audio_device_wave_win.cc
View File

@ -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

3
webrtc/modules/audio_device/win/audio_device_wave_win.h
View File

@ -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();

6
webrtc/system_wrappers/interface/event_wrapper.h
View File

@ -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

29
webrtc/system_wrappers/source/event.cc
View File

@ -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

14
webrtc/voice_engine/transmit_mixer.cc
View File

@ -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

5
webrtc/voice_engine/transmit_mixer.h
View File

@ -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

3
webrtc/voice_engine/voe_base_impl.cc
View File

@ -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

1
webrtc/voice_engine/voe_base_impl.h
View File

@ -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,

3
webrtc/voice_engine/voe_external_media_impl.cc
View File

@ -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();