Fixing an error in DebugDumpTest.

webrtc/modules/audio_processing/test/debug_dump_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
@@ skipping 223 matching lines @@
   // guaranteed if the debug dump is started on the first frame.
   void VerifyDebugDump(const std::string& dump_file_name);
 
  private:
   DebugDumpReplayer debug_dump_replayer_;
 };
 
 void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
 
-  if (const rtc::Optional<audioproc::Event> event =
+  while (const rtc::Optional<audioproc::Event> event =
       debug_dump_replayer_.GetNextEvent()) {
     debug_dump_replayer_.RunNextEvent();
     if (event->type() == audioproc::Event::STREAM) {
       const audioproc::Stream* msg = &event->stream();
       const StreamConfig output_config = debug_dump_replayer_.GetOutputConfig();
       const ChannelBuffer<float>* output = debug_dump_replayer_.GetOutput();
       // Check that output of APM is bit-exact to the output in the dump.
       ASSERT_EQ(output_config.num_channels(),
                 static_cast<size_t>(msg->output_channel_size()));
       ASSERT_EQ(output_config.num_frames() * sizeof(float),
@@ skipping 142 matching lines @@
   config.Set<ExperimentalNs>(new ExperimentalNs(true));
   DebugDumpGenerator generator(config);
   generator.StartRecording();
   generator.Process(100);
   generator.StopRecording();
   VerifyDebugDump(generator.dump_file_name());
 }
 
 }  // namespace test
 }  // namespace webrtc