| Index: webrtc/voice_engine/transmit_mixer.cc
|
| diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
|
| index 834510d8f697d9f992c198e0501c805027cda03f..a20f63b4295046cd1f2725d8d771a0b8d63d5f63 100644
|
| --- a/webrtc/voice_engine/transmit_mixer.cc
|
| +++ b/webrtc/voice_engine/transmit_mixer.cc
|
| @@ -34,7 +34,7 @@ TransmitMixer::OnPeriodicProcess()
|
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
|
| "TransmitMixer::OnPeriodicProcess()");
|
|
|
| -#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
|
| +#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| bool send_typing_noise_warning = false;
|
| bool typing_noise_detected = false;
|
| {
|
| @@ -191,7 +191,7 @@ TransmitMixer::TransmitMixer(uint32_t instanceId) :
|
| _fileRecording(false),
|
| _fileCallRecording(false),
|
| _audioLevel(),
|
| -#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| +#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| _typingNoiseWarningPending(false),
|
| _typingNoiseDetected(false),
|
| #endif
|
| @@ -342,7 +342,7 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
|
| AudioFrameOperations::SwapStereoChannels(&_audioFrame);
|
|
|
| // --- Annoying typing detection (utilizes the APM/VAD decision)
|
| -#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| +#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| TypingDetection(keyPressed);
|
| #endif
|
|
|
| @@ -1167,7 +1167,7 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
|
| _saturationWarning |= agc->stream_is_saturated();
|
| }
|
|
|
| -#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| +#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| void TransmitMixer::TypingDetection(bool keyPressed)
|
| {
|
| // We let the VAD determine if we're using this feature or not.
|
| @@ -1198,7 +1198,7 @@ int TransmitMixer::GetMixingFrequency()
|
| return _audioFrame.sample_rate_hz_;
|
| }
|
|
|
| -#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| +#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| int TransmitMixer::TimeSinceLastTyping(int &seconds)
|
| {
|
| // We check in VoEAudioProcessingImpl that this is only called when
|
| @@ -1208,7 +1208,7 @@ int TransmitMixer::TimeSinceLastTyping(int &seconds)
|
| }
|
| #endif
|
|
|
| -#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| +#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
| int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
|
| int costPerTyping,
|
| int reportingThreshold,
|
|
|