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..43804cb82689ab3497ea1e1c6ab231245600b0c9 100644
--- a/media/audio/fake_audio_input_stream.cc
+++ b/media/audio/fake_audio_input_stream.cc
@@ -185,7 +185,7 @@ void FakeAudioInputStream::DoCallback() {
     next_callback_time = base::TimeDelta();
 
   if (PlayingFromFile()) {
-    PlayFileLooping();
+    PlayFile();
   } else {
     PlayBeep();
   }
@@ -214,14 +214,16 @@ bool FakeAudioInputStream::PlayingFromFile() {
   return wav_audio_handler_.get() != nullptr;
 }
 
-void FakeAudioInputStream::PlayFileLooping() {
+void FakeAudioInputStream::PlayFile() {
+  // Stop playing if we've played out the whole file.

[phoglund_chromium, 2014/12/17 18:54:55]

The AGC test is the only user of this anyway so I can bluntly change it from
looping to not looping.

+  if (wav_audio_handler_->AtEnd(wav_file_read_pos_))
+    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);
 }