commit d45e078e894487b087e2cb636d41e00264b88fda
author Artem Titov <titovartem@webrtc.org> Tue Mar 27 08:15:26 2018
committer Commit Bot <commit-bot@chromium.org> Tue Mar 27 11:56:53 2018
tree 97dd7d7ce37a08288fdb101c90582da0e419715d
parent 959a1b93c773a6b120cd0fe3c89020c78792a555

Initialize variables to prevent usage of trash values, for exmaple, if NeedMorePlayData won't set the right value

Bug: webrtc:8946
Change-Id: I4eddcbdb5edfe004019356b9da7fff08ed2cbcdf
Reviewed-on: https://webrtc-review.googlesource.com/64882
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22624}

modules/audio_device/include/test_audio_device.cc

diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc
index b67d529..b674822 100644
--- a/modules/audio_device/include/test_audio_device.cc
+++ b/modules/audio_device/include/test_audio_device.cc
@@ -160,7 +160,7 @@
       if (capturing_) {
         // Capture 10ms of audio. 2 bytes per sample.
         const bool keep_capturing = capturer_->Capture(&recording_buffer_);
-        uint32_t new_mic_level;
+        uint32_t new_mic_level = 0;
         if (recording_buffer_.size() > 0) {
           audio_callback_->RecordedDataIsAvailable(
               recording_buffer_.data(), recording_buffer_.size(), 2,
@@ -173,9 +173,9 @@
         }
       }
       if (rendering_) {
-        size_t samples_out;
-        int64_t elapsed_time_ms;
-        int64_t ntp_time_ms;
+        size_t samples_out = 0;
+        int64_t elapsed_time_ms = 0;
+        int64_t ntp_time_ms = 0;
         const int sampling_frequency = renderer_->SamplingFrequency();
         audio_callback_->NeedMorePlayData(
             SamplesPerFrame(sampling_frequency), 2, renderer_->NumChannels(),