Add --quick flag to low bandwidth audio test

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <algorithm>
 
 #include "gflags/gflags.h"
 #include "webrtc/audio/test/low_bandwidth_audio_test.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 
 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.
 constexpr int kExtraRecordTimeMs = 500;
 
 std::string FileSampleRateSuffix() {
   return std::to_string(FLAGS_sample_rate_hz / 1000);
 }
 
 }  // namespace
@@ skipping 65 matching lines @@
   std::vector<AudioReceiveStream::Config>* receive_configs) {
   // Large bitrate by default.
   const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
                                               {{"stereo", "1"}});
   send_config->send_codec_spec =
       rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
           {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
 }
 
 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) {
+    // Let the recording run for a small amount of time to check if it works.
+    SleepMs(1000);
+  } 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() {
   const ::testing::TestInfo* const test_info =
       ::testing::UnitTest::GetInstance()->current_test_info();
 
   // Output information about the input and output audio files so that further
   // processing can be done by an external process.
   printf("TEST %s %s %s\n", test_info->name(),
          AudioInputFile().c_str(), AudioOutputFile().c_str());
@@ skipping 31 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc