Index: webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
index 3dce5c8bea68e787ede99a067e8af9298adff8f2..636698e9c10fc9da63c0f41583d043b88bee7e7b 100644 |
--- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
+++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
@@ -35,14 +35,14 @@ const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f, |
0.8608f, 0.8734f, 0.8861f, 0.8987f, |
0.9114f, 0.9241f, 0.9367f, 0.9494f, |
0.9620f, 0.9747f, 0.9873f, 1.0000f}; |
-const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]); |
+const size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]); |
} // namespace |
namespace webrtc { |
void CalculateEnergy(AudioFrame& audioFrame) |
{ |
audioFrame.energy_ = 0; |
- for(int position = 0; position < audioFrame.samples_per_channel_; |
+ for(size_t position = 0; position < audioFrame.samples_per_channel_; |
position++) |
{ |
// TODO(andrew): this can easily overflow. |
@@ -54,7 +54,7 @@ void CalculateEnergy(AudioFrame& audioFrame) |
void RampIn(AudioFrame& audioFrame) |
{ |
assert(rampSize <= audioFrame.samples_per_channel_); |
- for(int i = 0; i < rampSize; i++) |
+ for(size_t i = 0; i < rampSize; i++) |
{ |
audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] * |
audioFrame.data_[i]); |
@@ -64,9 +64,9 @@ void RampIn(AudioFrame& audioFrame) |
void RampOut(AudioFrame& audioFrame) |
{ |
assert(rampSize <= audioFrame.samples_per_channel_); |
- for(int i = 0; i < rampSize; i++) |
+ for(size_t i = 0; i < rampSize; i++) |
{ |
- const int rampPos = rampSize - 1 - i; |
+ const size_t rampPos = rampSize - 1 - i; |
audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] * |
audioFrame.data_[i]); |
} |