| 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]);
|
| }
|
|
|