Adding WebRTC Auto Gain Control Test (linux)

media/audio/fake_audio_input_stream.cc

Index: media/audio/fake_audio_input_stream.cc
diff --git a/media/audio/fake_audio_input_stream.cc b/media/audio/fake_audio_input_stream.cc
index 7fe3f825d28096114f6b4e867324765b1e8336e5..e0994e701cb7c219ba617ffab64eb5f3ecaeb50a 100644
--- a/media/audio/fake_audio_input_stream.cc
+++ b/media/audio/fake_audio_input_stream.cc
@@ -215,13 +215,16 @@ bool FakeAudioInputStream::PlayingFromFile() {
 }
 
 void FakeAudioInputStream::PlayFileLooping() {
+  // TODO(phoglund): reinstate looping/make optional after wrapping up AGC test.
+  if (wav_audio_handler_->AtEnd(wav_file_read_pos_))
+    //wav_file_read_pos_ = 0;
+    return;
+
   // Unfilled frames will be zeroed by CopyTo.
   size_t bytes_written;
   wav_audio_handler_->CopyTo(audio_bus_.get(), wav_file_read_pos_,
                              &bytes_written);
   wav_file_read_pos_ += bytes_written;
-  if (wav_audio_handler_->AtEnd(wav_file_read_pos_))
-    wav_file_read_pos_ = 0;
   callback_->OnData(this, audio_bus_.get(), buffer_size_, 1.0);
 }