media: Move audio_parameters and audio_point to media/base/

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/macros.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "build/build_config.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/mock_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.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"
 #include "media/base/audio_capturer_source.h"
+#include "media/base/audio_parameters.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::AtLeast;
 using ::testing::Return;
@@ skipping 542 matching lines @@
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source.get(), OnStop()).Times(1);
   capturer->Stop();
 
   // Even though this test don't use |capturer_source_| it will be stopped
   // during teardown of the test harness.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
 }
 
 }  // namespace content