1 files changed, 16 insertions, 25 deletions

diff --git a/chromium/third_party/webrtc/voice_engine/transmit_mixer.h b/chromium/third_party/webrtc/voice_engine/transmit_mixer.h
index d795eac265e..afeec3ac885 100644
--- a/chromium/third_party/webrtc/voice_engine/transmit_mixer.h
+++ b/chromium/third_party/webrtc/voice_engine/transmit_mixer.h
@@ -13,9 +13,11 @@
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_processing/typing_detection.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/utility/interface/file_player.h"
 #include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"
@@ -35,9 +37,7 @@ class MixedAudio;
 class Statistics;
 
 class TransmitMixer : public MonitorObserver,
-                      public FileCallback
-
-{
+                      public FileCallback {
 public:
     static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId);
 
@@ -70,6 +70,7 @@ public:
     // channels for encoding and sending to the network.
     void EncodeAndSend(const int voe_channels[], int number_of_voe_channels);
 
+    // Must be called on the same thread as PrepareDemux().
     uint32_t CaptureLevel() const;
 
     int32_t StopSend();
@@ -116,8 +117,6 @@ public:
 
     int IsPlayingFileAsMicrophone() const;
 
-    int ScaleFileAsMicrophonePlayout(float scale);
-
     int StartRecordingMicrophone(const char* fileName,
                                  const CodecInst* codecInst);
 
@@ -173,19 +172,20 @@ private:
     // sending codecs.
     void GetSendCodecInfo(int* max_sample_rate, int* max_channels);
 
-    int GenerateAudioFrame(const int16_t audioSamples[],
-                           int nSamples,
-                           int nChannels,
-                           int samplesPerSec);
+    void GenerateAudioFrame(const int16_t audioSamples[],
+                            int nSamples,
+                            int nChannels,
+                            int samplesPerSec);
     int32_t RecordAudioToFile(uint32_t mixingFrequency);
 
     int32_t MixOrReplaceAudioWithFile(
         int mixingFrequency);
 
-    void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level);
+    void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
+                      bool key_pressed);
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    int TypingDetection(bool keyPressed);
+    void TypingDetection(bool keyPressed);
 #endif
 
     // uses
@@ -198,7 +198,7 @@ private:
     // owns
     MonitorModule _monitorModule;
     AudioFrame _audioFrame;
-    PushResampler resampler_;  // ADM sample rate -> mixing rate
+    PushResampler<int16_t> resampler_;  // ADM sample rate -> mixing rate
     FilePlayer* _filePlayerPtr;
     FileRecorder* _fileRecorderPtr;
     FileRecorder* _fileCallRecorderPtr;
@@ -214,19 +214,9 @@ private:
     CriticalSectionWrapper& _callbackCritSect;
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    int32_t _timeActive;
-    int32_t _timeSinceLastTyping;
-    int32_t _penaltyCounter;
+    webrtc::TypingDetection _typingDetection;
     bool _typingNoiseWarningPending;
     bool _typingNoiseDetected;
-
-    // Tunable treshold values
-    int _timeWindow; // nr of10ms slots accepted to count as a hit.
-    int _costPerTyping; // Penalty added for a typing + activity coincide.
-    int _reportingThreshold; // Threshold for _penaltyCounter.
-    int _penaltyDecay; // How much we reduce _penaltyCounter every 10 ms.
-    int _typeEventDelay; // How old typing events we allow
-
 #endif
     bool _saturationWarning;
 
@@ -239,10 +229,11 @@ private:
     int32_t _remainingMuteMicTimeMs;
     bool stereo_codec_;
     bool swap_stereo_channels_;
+    scoped_ptr<int16_t[]> mono_buffer_;
 };
 
-#endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
-
 }  // namespace voe
 
 }  // namespace webrtc
+
+#endif  // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H