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, |