commit 023cc5abc7d25fb3133b4d0206b67dcc6204b6e8
author andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Sat Jan 11 01:25:53 2014
committer andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Sat Jan 11 01:25:53 2014
tree fe1d366c3d22cdec120438addde38f3d509dc780
parent 573a1b45b5b7638605d9727be57c73e838d6ee45

Minor voice engine improvements around AGC.

- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.

R=aluebs@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/voice_engine/transmit_mixer.cc

diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index b9618c8..e8b3365 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -494,7 +494,6 @@
 
 uint32_t TransmitMixer::CaptureLevel() const
 {
-    CriticalSectionScoped cs(&_critSect);
     return _captureLevel;
 }
 
@@ -1341,11 +1340,10 @@
     assert(false);
   }
 
-  CriticalSectionScoped cs(&_critSect);
-
   // Store new capture level. Only updated when analog AGC is enabled.
   _captureLevel = agc->stream_analog_level();
 
+  CriticalSectionScoped cs(&_critSect);
   // Triggers a callback in OnPeriodicProcess().
   _saturationWarning |= agc->stream_is_saturated();
 }

webrtc/voice_engine/transmit_mixer.h

diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 0f70312..dc46cf7 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -70,6 +70,7 @@
     // 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();

webrtc/voice_engine/voe_base_impl.cc

diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 682a4a5..9434863 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -1150,17 +1150,11 @@
   assert(_shared->transmit_mixer() != NULL);
   assert(_shared->audio_device() != NULL);
 
-  bool is_analog_agc(false);
-  if (_shared->audio_processing() &&
-      _shared->audio_processing()->gain_control()->mode() ==
-          GainControl::kAdaptiveAnalog) {
-    is_analog_agc = true;
-  }
-
-  // Only deal with the volume in adaptive analog mode.
   uint32_t max_volume = 0;
   uint16_t current_voe_mic_level = 0;
-  if (is_analog_agc) {
+  // Check for zero to skip this calculation; the consumer may use this to
+  // indicate no volume is available.
+  if (current_volume != 0) {
     // Scale from ADM to VoE level range
     if (_shared->audio_device()->MaxMicrophoneVolume(&max_volume) == 0) {
       if (max_volume) {
@@ -1209,9 +1203,6 @@
                                              number_of_voe_channels);
   }
 
-  if (!is_analog_agc)
-    return 0;
-
   // Scale from VoE to ADM level range.
   uint32_t new_voe_mic_level = _shared->transmit_mixer()->CaptureLevel();