Prepare to roll Chromium to 149181.

- This roll brings in VS2010 by default. The buildbots
  need updating (issue710).
- We'll roll to 149181 later (past current Canary) to fix
  a Mac gyp issue:
  https://chromiumcodereview.appspot.com/10824105
- Chromium is now using a later libvpx than us. We should
  investigate rolling our standalone build.
- Fix set-but-unused-warning
- Fix -Wunused-private-field warnings on Mac.

TBR=kjellander@webrtc.org
BUG=issue709,issue710
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/709007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_coding/main/test/APITest.cc

@@ -81,8 +81,6 @@ _processEventB(NULL),
 _apiEventB(NULL),
 _codecCntrA(0),
 _codecCntrB(0),
-_testCntrA(1),
-_testCntrB(1),
 _thereIsEncoderA(false),
 _thereIsEncoderB(false),
 _thereIsDecoderA(false),

src/modules/audio_coding/main/test/APITest.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -123,10 +123,6 @@ private:
     WebRtc_UWord8 _codecCntrA;
     WebRtc_UWord8 _codecCntrB;
 
-    // keep track of tests
-    WebRtc_UWord8 _testCntrA;
-    WebRtc_UWord8 _testCntrB;
-
     // Is set to true if there is no encoder in either side
     bool _thereIsEncoderA;
     bool _thereIsEncoderB;

src/modules/audio_coding/main/test/Channel.cc

@@ -247,7 +247,6 @@ _leftChannel(true),
 _lastInTimestamp(0),
 _packetLoss(0),
 _useFECTestWithPacketLoss(false),
-_chID(chID),
 _beginTime(TickTime::MillisecondTimestamp()),
 _totalBytes(0)
 {

src/modules/audio_coding/main/test/Channel.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -115,7 +115,6 @@ private:
     // FEC Test variables
     WebRtc_Word16           _packetLoss;
     bool                    _useFECTestWithPacketLoss;
-    WebRtc_Word16           _chID;
     WebRtc_UWord64          _beginTime;
     WebRtc_UWord64          _totalBytes;
 };

src/modules/audio_coding/main/test/EncodeDecodeTest.cc

@@ -50,8 +50,6 @@ Sender::Sender()
     : _acm(NULL),
       _pcmFile(),
       _audioFrame(),
-      _payloadSize(0),
-      _timeStamp(0),
       _packetization(NULL) {
 }
 

src/modules/audio_coding/main/test/EncodeDecodeTest.h

@@ -61,8 +61,6 @@ class Sender {
   AudioCodingModule* _acm;
   PCMFile _pcmFile;
   AudioFrame _audioFrame;
-  WebRtc_UWord16 _payloadSize;
-  WebRtc_UWord32 _timeStamp;
   TestPacketization* _packetization;
 };
 
@@ -81,7 +79,6 @@ class Receiver {
 
  private:
   AudioCodingModule* _acm;
-  bool _rtpEOF;
   RTPStream* _rtpStream;
   PCMFile _pcmFile;
   WebRtc_Word16* _playoutBuffer;

src/modules/audio_coding/main/test/TestStereo.h

@@ -48,7 +48,6 @@ class TestPackStereo : public AudioPacketizationCallback {
  private:
   AudioCodingModule* receiver_acm_;
   WebRtc_Word16 seq_no_;
-  WebRtc_UWord8 payload_data_[60 * 32 * 2 * 2];
   WebRtc_UWord32 timestamp_diff_;
   WebRtc_UWord32 last_in_timestamp_;
   WebRtc_UWord64 total_bytes_;

src/modules/audio_coding/main/test/TwoWayCommunication.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -53,9 +53,6 @@ private:
     PCMFile _outFileRefA;
     PCMFile _outFileRefB;
 
-    DTMFDetector* _dtmfDetectorA;
-    DTMFDetector* _dtmfDetectorB;
-
     int _testMode;
 };
 

src/modules/audio_device/main/source/mac/audio_device_mac.h

@@ -355,8 +355,6 @@ private:
     bool _doStopRec; // For rec if not shared device
     bool _macBookPro;
     bool _macBookProPanRight;
-    bool _stereoRender;
-    bool _stereoRenderRequested;
 
     AudioConverterRef _captureConverter;
     AudioConverterRef _renderConverter;
@@ -376,7 +374,6 @@ private:
     WebRtc_Word32 _renderDelayOffsetSamples;
 
 private:
-    WebRtc_UWord16 _playBufDelay; // playback delay
     WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
 
     WebRtc_UWord16 _playWarning;

src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc

@@ -18,8 +18,7 @@ namespace webrtc
 
 AudioDeviceUtilityMac::AudioDeviceUtilityMac(const WebRtc_Word32 id) :
     _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
+    _id(id)
-    _lastError(AudioDeviceModule::kAdmErrNone)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
                  "%s created", __FUNCTION__);

src/modules/audio_device/main/source/mac/audio_device_utility_mac.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -29,7 +29,6 @@ public:
 private:
     CriticalSectionWrapper& _critSect;
     WebRtc_Word32 _id;
-    AudioDeviceModule::ErrorCode _lastError;
 };
 
 } //  namespace webrtc

src/modules/audio_device/main/test/audio_device_test_api.cc

@@ -76,8 +76,7 @@ class AudioEventObserverAPI: public AudioDeviceObserver {
 class AudioTransportAPI: public AudioTransport {
  public:
   AudioTransportAPI(AudioDeviceModule* audioDevice)
-      : audio_device_(audioDevice),
+      : rec_count_(0),
-        rec_count_(0),
         play_count_(0) {
   }
 
@@ -129,7 +128,6 @@ class AudioTransportAPI: public AudioTransport {
   }
 
  private:
-  AudioDeviceModule* audio_device_;
   WebRtc_UWord32 rec_count_;
   WebRtc_UWord32 play_count_;
 };

src/modules/audio_device/main/test/func_test_manager.cc

@@ -68,8 +68,7 @@ const char* GetResource(const char* resource)
 namespace webrtc
 {
 
-AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice) :
+AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice)
-    _audioDevice(audioDevice)
 {
 }
 

src/modules/audio_device/main/test/func_test_manager.h

@@ -85,8 +85,6 @@ public:
 public:
     ErrorCode _error;
     WarningCode _warning;
-private:
-    AudioDeviceModule* _audioDevice;
 };
 
 // ----------------------------------------------------------------------------

src/modules/video_coding/codecs/test_framework/normal_async_test.h

@@ -38,8 +38,7 @@ class FrameQueue
 public:
     FrameQueue()
     :
-        _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock()),
+        _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock())
-        _prevTS(-1)
     {
     }
 
@@ -56,7 +55,6 @@ public:
 private:
     webrtc::RWLockWrapper&                       _queueRWLock;
     std::queue<FrameQueueTuple *>     _frameBufferQueue;
-    WebRtc_Word64                       _prevTS;
 };
 
 // feedback signal to encoder

src/modules/video_coding/codecs/test_framework/unit_test.h

@@ -77,8 +77,6 @@ public:
                                    WebRtc_UWord32 decoderSpecificSize = 0,
                                    void* decoderSpecificInfo = NULL) :
       _encodedVideoBuffer(buffer),
-      _decoderSpecificInfo(decoderSpecificInfo),
-      _decoderSpecificSize(decoderSpecificSize),
       _encodeComplete(false) {}
     WebRtc_Word32 Encoded(webrtc::EncodedImage& encodedImage,
                           const webrtc::CodecSpecificInfo* codecSpecificInfo,
@@ -89,8 +87,6 @@ public:
     webrtc::VideoFrameType EncodedFrameType() const;
 private:
     TestVideoEncodedBuffer* _encodedVideoBuffer;
-    void* _decoderSpecificInfo;
-    WebRtc_UWord32 _decoderSpecificSize;
     bool _encodeComplete;
     webrtc::VideoFrameType _encodedFrameType;
 };

src/modules/video_coding/main/test/normal_test.h

@@ -52,7 +52,6 @@ private:
     WebRtc_UWord32              _skipCnt;
     webrtc::VideoCodingModule*  _VCMReceiver;
     webrtc::FrameType           _frameType;
-    WebRtc_UWord8*              _payloadData; // max payload size??
     WebRtc_UWord16              _seqNo;
     NormalTest&                 _test;
 }; // end of VCMEncodeCompleteCallback

src/modules/video_coding/main/test/test_callbacks.h

@@ -78,7 +78,6 @@ private:
     float              _encodedBytes;
     VideoCodingModule* _VCMReceiver;
     FrameType          _frameType;
-    WebRtc_UWord8*     _payloadData;
     WebRtc_UWord16     _seqNo;
     bool               _encodeComplete;
     WebRtc_Word32      _width;
@@ -94,7 +93,6 @@ class VCMRTPEncodeCompleteCallback: public VCMPacketizationCallback
 public:
     VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) :
         _encodedBytes(0),
-        _seqNo(0),
         _encodeComplete(false),
         _RTPModule(rtp) {}
 
@@ -128,8 +126,6 @@ public:
 private:
     float              _encodedBytes;
     FrameType          _frameType;
-    WebRtc_UWord8*     _payloadData;
-    WebRtc_UWord16     _seqNo;
     bool               _encodeComplete;
     RtpRtcp*           _RTPModule;
     WebRtc_Word16      _width;

src/modules/video_render/main/source/mac/video_render_nsopengl.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -87,8 +87,6 @@ private:
     int _stretchedHeight;
     int _oldStretchedHeight;
     int _oldStretchedWidth;
-    int _xOldWidth;
-    int _yOldHeight;
     unsigned char* _buffer;
     int _bufferSize;
     int _incommingBufferSize;

src/modules/video_render/main/source/mac/video_render_nsopengl.mm

@@ -34,8 +34,6 @@ _stretchedWidth( 0),
 _stretchedHeight( 0),
 _oldStretchedHeight( 0),
 _oldStretchedWidth( 0),
-_xOldWidth( 0),
-_yOldHeight( 0),
 _buffer( 0),
 _bufferSize( 0),
 _incommingBufferSize( 0),

src/test/testsupport/frame_writer.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -60,7 +60,6 @@ class FrameWriterImpl : public FrameWriter {
  private:
   std::string output_filename_;
   int frame_length_in_bytes_;
-  int number_of_frames_;
   FILE* output_file_;
 };
 

src/video_engine/main/test/WindowsTest/windowstest.gypi

@@ -8,7 +8,9 @@
 
 {
   'conditions': [
-    ['OS=="win"', {
+    # TODO(kjellander): Support UseoFMFC on VS2010.
+    # http://code.google.com/p/webrtc/issues/detail?id=709
+    ['OS=="win" and MSVS_VERSION < "2010"', {
       'targets': [
         # WinTest - GUI test for Windows
         {

src/video_engine/stream_synchronization_unittest.cc

@@ -203,7 +203,6 @@ TEST_F(StreamSynchronizationTest, AudioDelay) {
   int current_audio_delay_ms = 0;
   int delay_ms = 200;
   int extra_audio_delay_ms = 0;
-  int current_extra_delay_ms = 0;
   int total_video_delay_ms = 0;
 
   EXPECT_EQ(0, DelayedVideo(delay_ms, current_audio_delay_ms,
@@ -212,7 +211,7 @@ TEST_F(StreamSynchronizationTest, AudioDelay) {
   // The audio delay is not allowed to change more than this in 1 second.
   EXPECT_EQ(kMaxAudioDiffMs, extra_audio_delay_ms);
   current_audio_delay_ms = extra_audio_delay_ms;
-  current_extra_delay_ms = extra_audio_delay_ms;
+  int current_extra_delay_ms = extra_audio_delay_ms;
 
   send_time_->IncreaseTimeMs(1000);
   receive_time_->IncreaseTimeMs(800);
@@ -273,7 +272,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) {
   int audio_delay_ms = 100;
   int video_delay_ms = 300;
   int extra_audio_delay_ms = 0;
-  int current_extra_delay_ms = 0;
   int total_video_delay_ms = 0;
 
   EXPECT_EQ(0, DelayedAudioAndVideo(audio_delay_ms,
@@ -285,7 +283,7 @@ TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) {
   // The audio delay is not allowed to change more than this in 1 second.
   EXPECT_EQ(kMaxAudioDiffMs, extra_audio_delay_ms);
   current_audio_delay_ms = extra_audio_delay_ms;
-  current_extra_delay_ms = extra_audio_delay_ms;
+  int current_extra_delay_ms = extra_audio_delay_ms;
 
   send_time_->IncreaseTimeMs(1000);
   receive_time_->IncreaseTimeMs(800);
@@ -358,7 +356,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) {
   int audio_delay_ms = 300;
   int video_delay_ms = 100;
   int extra_audio_delay_ms = 0;
-  int current_extra_delay_ms = 0;
   int total_video_delay_ms = 0;
 
   EXPECT_EQ(0, DelayedAudioAndVideo(audio_delay_ms,
@@ -369,7 +366,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) {
   EXPECT_EQ(kMaxVideoDiffMs, total_video_delay_ms);
   EXPECT_EQ(0, extra_audio_delay_ms);
   current_audio_delay_ms = extra_audio_delay_ms;
-  current_extra_delay_ms = extra_audio_delay_ms;
 
   send_time_->IncreaseTimeMs(1000);
   receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms,
@@ -384,7 +380,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) {
   EXPECT_EQ(2 * kMaxVideoDiffMs, total_video_delay_ms);
   EXPECT_EQ(0, extra_audio_delay_ms);
   current_audio_delay_ms = extra_audio_delay_ms;
-  current_extra_delay_ms = extra_audio_delay_ms;
 
   send_time_->IncreaseTimeMs(1000);
   receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms,
@@ -398,7 +393,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) {
                                     &total_video_delay_ms));
   EXPECT_EQ(audio_delay_ms - video_delay_ms, total_video_delay_ms);
   EXPECT_EQ(0, extra_audio_delay_ms);
-  current_extra_delay_ms = extra_audio_delay_ms;
 
   // Simulate that NetEQ introduces some audio delay.
   current_audio_delay_ms = 50;
@@ -415,7 +409,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) {
   EXPECT_EQ(audio_delay_ms - video_delay_ms + current_audio_delay_ms,
             total_video_delay_ms);
   EXPECT_EQ(0, extra_audio_delay_ms);
-  current_extra_delay_ms = extra_audio_delay_ms;
 
   // Simulate that NetEQ reduces its delay.
   current_audio_delay_ms = 10;

src/video_engine/test/auto_test/vie_auto_test.gypi

@@ -99,22 +99,15 @@
         'source/vie_window_manager_factory_win.cc',
       ],
       'conditions': [
-        # TODO(andrew): this likely isn't an actual dependency. It should be
-        # included in webrtc.gyp or video_engine.gyp instead.
         ['OS=="android"', {
           'libraries': [
             '-lGLESv2',
             '-llog',
           ],
         }],
-        ['OS=="win"', {
-          'dependencies': [
-            'vie_win_test',
-          ],
-        }],
         ['OS=="linux"', {
-          # TODO(andrew): these should be provided directly by the projects
+          # TODO(andrew): These should be provided directly by the projects
-          #   # which require them instead.
+          #               which require them instead.
           'libraries': [
             '-lXext',
             '-lX11',

src/video_engine/test/libvietest/include/vie_file_capture_device.h

@@ -50,7 +50,6 @@ class ViEFileCaptureDevice {
   webrtc::CriticalSectionWrapper* mutex_;
 
   WebRtc_UWord32 frame_length_;
-  WebRtc_UWord8* frame_buffer_;
   WebRtc_UWord32 width_;
   WebRtc_UWord32 height_;
 };

src/voice_engine/test/voice_engine_tests.gypi

@@ -104,7 +104,9 @@
     },
   ],
   'conditions': [
-    ['OS=="win"', {
+    # TODO(kjellander): Support UseoFMFC on VS2010.
+    # http://code.google.com/p/webrtc/issues/detail?id=709
+    ['OS=="win" and MSVS_VERSION < "2010"', {
       'targets': [
         # WinTest - GUI test for Windows
         {