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Prepare to roll Chromium to 149181.

Browse Source 
- 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
This commit is contained in:
andrew@webrtc.org
2012-08-01 01:40:02 +00:00
parent bf853918eb
commit d7a71d0719
26 changed files with 145 additions and 199 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
src/modules/audio_coding/main/test/APITest.cc
View File

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

6
src/modules/audio_coding/main/test/APITest.h
View File

@@ -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;

1
src/modules/audio_coding/main/test/Channel.cc
View File

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

3
src/modules/audio_coding/main/test/Channel.h
View File

@@ -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;
};

2
src/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

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

3
src/modules/audio_coding/main/test/EncodeDecodeTest.h
View File

@@ -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;

1
src/modules/audio_coding/main/test/TestStereo.h
View File

@@ -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_;

5
src/modules/audio_coding/main/test/TwoWayCommunication.h
View File

@@ -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;
};

3
src/modules/audio_device/main/source/mac/audio_device_mac.h
View File

@@ -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;

3
src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc
View File

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

3
src/modules/audio_device/main/source/mac/audio_device_utility_mac.h
View File

@@ -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

4
src/modules/audio_device/main/test/audio_device_test_api.cc
View File

@@ -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_;
};

3
src/modules/audio_device/main/test/func_test_manager.cc
View File

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

2
src/modules/audio_device/main/test/func_test_manager.h
View File

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

4
src/modules/video_coding/codecs/test_framework/normal_async_test.h
View File

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

4
src/modules/video_coding/codecs/test_framework/unit_test.h
View File

@@ -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;
};

1
src/modules/video_coding/main/test/normal_test.h
View File

@@ -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

4
src/modules/video_coding/main/test/test_callbacks.h
View File

@@ -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;

4
src/modules/video_render/main/source/mac/video_render_nsopengl.h
View File

@@ -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;

2
src/modules/video_render/main/source/mac/video_render_nsopengl.mm
View File

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

3
src/test/testsupport/frame_writer.h
View File

@@ -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_;
};

4
src/video_engine/main/test/WindowsTest/windowstest.gypi
View File

@@ -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
{

11
src/video_engine/stream_synchronization_unittest.cc
View File

@@ -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;

11
src/video_engine/test/auto_test/vie_auto_test.gypi
View File

@@ -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
# # which require them instead.
# TODO(andrew): These should be provided directly by the projects
# which require them instead.
'libraries': [
'-lXext',
'-lX11',

1
src/video_engine/test/libvietest/include/vie_file_capture_device.h
View File

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

4
src/voice_engine/test/voice_engine_tests.gypi
View File

@@ -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
{