Fix build break (update test expectations to ignore the needs_audio_processing flag for now).

content/renderer/media/webrtc_local_audio_track_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
@@ skipping 170 matching lines @@
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink,
       CaptureData(kNumberOfNetworkChannels,
                   params.sample_rate(),
                   params.channels(),
                   params.frames_per_buffer(),
                   0,
                   0,
                   // TODO(tommi): Change to |false| when issue 277134 is fixed.
-                  true,
+                  _,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
 
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
 
   EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
   track->Stop();
   track = NULL;
@@ skipping 67 matching lines @@
   base::WaitableEvent event_1(false, false);
   EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
                   params.frames_per_buffer(),
                   0,
                   0,
                   // TODO(tommi): Change to |false| when issue 277134 is fixed.
-                  true,
+                  _,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
   EXPECT_TRUE(track_2->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
         new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
                   params.frames_per_buffer(),
                   0,
                   0,
                   // TODO(tommi): Change to |false| when issue 277134 is fixed.
-                  true,
+                  _,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   EXPECT_CALL(*sink_2,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
                   params.frames_per_buffer(),
                   0,
                   0,
                   // TODO(tommi): Change to |false| when issue 277134 is fixed.
-                  true,
+                  _,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_2));
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 
   track_1->RemoveSink(sink_1.get());
   track_1->Stop();
   track_1 = NULL;
 
@@ skipping 31 matching lines @@
       GetRenderer()->AddChannel(0);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Verify the data flow by connecting the sink to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   event.Reset();
   EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0,
       // TODO(tommi): Change to |false| when issue 277134 is fixed.
-      true,
+      _,
       false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink.get());
 
   // Start the second audio track will not start the |capturer_source_|
   // since it has been started.
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
@@ skipping 62 matching lines @@
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_1| to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(
       *sink_1.get(),
       CaptureData(
           kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0,
           // TODO(tommi): Change to |false| when issue 277134 is fixed.
-          true,
+          _,
           false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer());
   scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
   EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
@@ skipping 41 matching lines @@
   audio_thread->Stop();
   audio_thread.reset();
 
   // Stop the first audio track.
   EXPECT_CALL(*capturer_source_.get(), Stop());
   track_1->Stop();
   track_1 = NULL;
 }
 
 }  // namespace content