Index: talk/app/webrtc/test/fakeaudiocapturemodule.cc |
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.cc b/talk/app/webrtc/test/fakeaudiocapturemodule.cc |
index c6339d3c3f4295d88b6b45abc41cb55a8943fe11..cf645a5c76faafde7ebdc13c354963736530ce2c 100644 |
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.cc |
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.cc |
@@ -623,8 +623,8 @@ bool FakeAudioCaptureModule::Initialize() { |
void FakeAudioCaptureModule::SetSendBuffer(int value) { |
Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_); |
- const int buffer_size_in_samples = sizeof(send_buffer_) / |
- kNumberBytesPerSample; |
+ const int buffer_size_in_samples = |
+ sizeof(send_buffer_) / kNumberBytesPerSample; |
for (int i = 0; i < buffer_size_in_samples; ++i) { |
buffer_ptr[i] = value; |
} |
@@ -636,8 +636,8 @@ void FakeAudioCaptureModule::ResetRecBuffer() { |
bool FakeAudioCaptureModule::CheckRecBuffer(int value) { |
const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_); |
- const int buffer_size_in_samples = sizeof(rec_buffer_) / |
- kNumberBytesPerSample; |
+ const int buffer_size_in_samples = |
+ sizeof(rec_buffer_) / kNumberBytesPerSample; |
for (int i = 0; i < buffer_size_in_samples; ++i) { |
if (buffer_ptr[i] >= value) return true; |
} |