Index: webrtc/api/test/fakeaudiocapturemodule.cc |
diff --git a/webrtc/api/test/fakeaudiocapturemodule.cc b/webrtc/api/test/fakeaudiocapturemodule.cc |
index a32ef64d03772f324dea53eb2baabe95f2c86442..43ff664d0c12c5ad5267a04d1cc7226d7305cc6a 100644 |
--- a/webrtc/api/test/fakeaudiocapturemodule.cc |
+++ b/webrtc/api/test/fakeaudiocapturemodule.cc |
@@ -627,7 +627,7 @@ void FakeAudioCaptureModule::UpdateProcessing(bool start) { |
process_thread_.reset(new rtc::Thread()); |
process_thread_->Start(); |
} |
- process_thread_->Post(this, MSG_START_PROCESS); |
+ process_thread_->Post(RTC_FROM_HERE, this, MSG_START_PROCESS); |
} else { |
if (process_thread_) { |
process_thread_->Stop(); |
@@ -668,7 +668,7 @@ void FakeAudioCaptureModule::ProcessFrameP() { |
const int64_t current_time = rtc::TimeMillis(); |
const int64_t wait_time = |
(next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0; |
- process_thread_->PostDelayed(wait_time, this, MSG_RUN_PROCESS); |
+ process_thread_->PostDelayed(RTC_FROM_HERE, wait_time, this, MSG_RUN_PROCESS); |
} |
void FakeAudioCaptureModule::ReceiveFrameP() { |