Add --quick flag to low bandwidth audio test

webrtc/audio/test/low_bandwidth_audio_test.cc

Index: webrtc/audio/test/low_bandwidth_audio_test.cc
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc
index f9e6e2b9d1fdaab12f0bfdc7ccfec7f8b486df26..4ddc215883d08873b7b89167d51b2c9bc77871eb 100644
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc
@@ -21,6 +21,10 @@
 DEFINE_int32(sample_rate_hz, 16000,
              "Sample rate (Hz) of the produced audio files.");
 
+DEFINE_bool(quick, false,
+            "Don't do the full audio recording. "
+            "Used to quickly check that the test runs without crashing.");
+
 namespace {
 
 // Wait half a second between stopping sending and stopping receiving audio.
@@ -106,10 +110,14 @@ void AudioQualityTest::ModifyAudioConfigs(
 }
 
 void AudioQualityTest::PerformTest() {
-  // Wait until the input audio file is done...
-  send_audio_device_->WaitForRecordingEnd();
-  // and some extra time to account for network delay.
-  SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
+  if (FLAGS_quick) {
+    SleepMs(1000);

[kwiberg-webrtc, 2017/05/04 09:30:14]

Maybe a comment here explaining what the wait is good for?

+  } else {
+    // Wait until the input audio file is done...
+    send_audio_device_->WaitForRecordingEnd();
+    // and some extra time to account for network delay.
+    SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
+  }
 }
 
 void AudioQualityTest::OnTestFinished() {