ebdb0e3ad0
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
...
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.
This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.
BUG=
TBR=solenberg@webrtc.org ,henrikg@webrtc.org ,stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 23:49:08 +00:00
45846977f9
Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted.
...
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 15:46:46 +00:00
ed865b5d46
NetEq4: Changing the behavior of playout_timestamp_ update
...
The variable playout_timestamp_ was not updated to the latest decoded
timestamp while comfort noise was played. Instead, it was upadted using
dead reckoning, which caused it to drift away from the timestamps of the
incoming CNG packets. Now it is updated also during comfort noise
playout.
Since the change is only in NetEq4, this change also makes the test
PlaysOutAudioAndVideoInSync use both ACM1/NetEq3 and ACM2/NetEq4.
Re-enabling one NetEq unit test that is no longer failing thanks to this CL.
BUG=2932
R=stefan@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 10:28:07 +00:00
845862f279
Adding a new ramp-up-down-up test
...
The new test is based upon the exisiting rampup test, but also adds
a low-rate period. The main purpose of the test is to verify the
SuspendBelowMinBitrate functionality, which must be enabled for the
test to pass.
The CL also adds a change to the min bitrate in the send-side bandwidth
estimator when SuspendBelowMinBitrate is enabled.
An anonymous namespace is added around the StreamObserver classes
in the test to avoid silent linker conflicts that could happen
otherwise.
Note: this CL depends on https://webrtc-codereview.appspot.com/9049004/
BUG=2636
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 07:19:28 +00:00
a0d11da359
Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this.
...
BUG=2990
TEST=Manually adding a high number without any noticable change.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-05 15:18:45 +00:00
3e0b60f465
Switch to correct interpretation of int and float input data in audio_processing_unittest
...
BUG=N/A
TESTED=trybots
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5642 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-05 00:18:53 +00:00
17e40641b3
Add a deinterleaved float interface to AudioProcessing.
...
This is mainly to support the native audio format in Chrome. Although
this implementation just moves the float->int conversion under the hood,
we will transition AudioProcessing towards supporting this format
throughout.
- Add a test which verifies we get identical output with the float and
int interfaces.
- The float and int wrappers are tasked with conversion to the
AudioBuffer format. A new shared Process/Analyze method does most of
the work.
- Add a new field to the debug.proto to hold deinterleaved data.
- Add helpers to audio_utils.cc, and start using numeric_limits.
- Note that there was no performance difference between numeric_limits
and a literal value when measured on Linux using gcc or clang.
BUG=2894
R=aluebs@webrtc.org , bjornv@webrtc.org , henrikg@webrtc.org , tommi@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-04 20:58:13 +00:00
7bd4a27502
VideoCaptureAndroid: don't deliver frames after stopCapture().
...
Because stopCapture() and onPreviewFrame() are called on different threads, and
are both synchronized, it's possible for onPreviewFrame() to commence execution
after stopCapture() has completed, causing a SEGV because the native code is no
longer prepared to accept frames.
Clarify the contract around synchronized methods in this class to hopefully
avoid similar bugs in future.
BUG=2947
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5639 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-04 18:17:55 +00:00
be50ab645a
Including algorithm header to avoid VS2013 breakage
...
The header file <algorithm> must be included when std::min and std::max
are used.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-04 15:10:03 +00:00
0117d1c48c
Fix compilation errors under clang 3.5.
...
Enables building tip-of-tree clang which introduces new warnings that
cause compilation errors in our code base (-Werror).
BUG=
R=andrew@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 16:47:03 +00:00
9fd8d87ff5
Adds APIs for reporting pacer queuing delay.
...
BUG=2775
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8959005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27 22:32:40 +00:00
56e4a05053
Remove ProcessingComponent's dependence on AudioProcessingImpl.
...
- Move needed accessors to AudioProcessing.
- Inject the crit directly as a dependency.
- Remove the now unneeded EchoCancellationImplWrapper.
BUG=2894
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5620 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27 22:23:17 +00:00
f0fc72f70e
Call PrintWindow for the first time of capturing to capture the window frames correctly.
...
This will fix artifacts on the captured window frames, especially for cmd, which
sometimes leaks glimpss of other window's content.
BUG=
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27 16:43:12 +00:00
0231e801d6
Invalidate the whole screen when the frame size is changed.
...
Otherwise we'll compare frames of different sizes and read into invalid
memory.
BUG=https://code.google.com/p/chromium/issues/detail?id=345498
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/9149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5614 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-26 18:54:57 +00:00
bc1d22461b
Add experimental noise suppression flag to audioproc test
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-25 16:50:22 +00:00
7f52a6ef2b
Split the implementation of VP8Encoder|Decoder::Create into a seperated file
...
(vp8_factory.cc).
R=fischman@webrtc.org , marpan@google.com , marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24 23:56:39 +00:00
4f0801bd39
AviRecorder is missing a critical section.
...
BUG=2885
TEST=AUTOTEST
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24 09:19:36 +00:00
33af96c5c2
Removed unused mock methods in audio_processing
...
TESTED=trybots,modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8999005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5597 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 23:56:05 +00:00
0f2809a5ac
Add RTCP packet class.
...
Adds packet types: sr, rr, bye, fir.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 08:14:45 +00:00
c0907eff42
MIPS optimizations for AEC audio processing module
...
The resulting output streams obtained by testing with audioproc test application
are bit-exact with generic C code output streams.
Performance gain achieved:
- mips32 ~ 17%
- mips32r2 ~ 20%
- mipsdsp & mipsdspr2 ~ 21%
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7359004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 00:13:31 +00:00
d617a44a4f
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
...
- Transition scoped_ptr_mallocs to scoped_ptr.
- AlignedFreeDeleter matches Chromium's version.
TESTED=try bots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8969005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 21:08:36 +00:00
04a691adac
Removing a variable that was never read
...
In NetEq4, the local variable discard_count in
PacketBuffer::DiscardOldPackets() was incremented but never read.
Removing it.
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 15:27:00 +00:00
66061992fb
ifdef the alsa code based on macro USE_X11
...
BUG=none
TEST=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 03:05:05 +00:00
78f0db4710
Fix the break caused by r5579.
...
TBR=tlegrand@google.com
BUG=
Review URL: https://webrtc-codereview.appspot.com/8939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 23:07:31 +00:00
c2d69d3229
Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable.
...
BUG=2944
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 20:31:17 +00:00
97e7a640d8
Make WindowCapturerLinux handling window resize events.
...
We need to re-initialize the XServerPixelBuffer to the new size
when a window resize event is received.
BUG=https://code.google.com/p/chromium/issues/detail?id=339953
R=sergeyu@chromium.org , wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 17:28:41 +00:00
056287eee0
This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both.
...
BUG=issue2874
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 13:45:54 +00:00
8098e07478
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
...
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().
BUG=2638
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 11:59:02 +00:00
e384104166
Fix DesktopAndCursorComposer not to crash
...
DesktopAndCursorComposer was crashing when screen/window
capturer returns a NULL frame due to an error.
BUG=crbug.com/344093
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 23:26:34 +00:00
27c6980239
Move the volume quantization workaround from VoE to AGC.
...
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.
Add a test to verify the behavior.
TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 20:24:56 +00:00
c320027d6a
Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called
...
twice with the same settings.
Without this change, setting up a call with the new video API will
print a trace warning.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:51:00 +00:00
2086e0fbf3
Remove unnecessary warnings.
...
BUG=
TEST=try job
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8719005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:22:20 +00:00
346094cb01
Incorrect overhead calculation when using FEC + RTP extension headers.
...
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.
BUG=2899
R=phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:40:33 +00:00
340746aa13
Misc small nits in NetEq
...
Fixing a few small things found recently. This is mostly cosmetics.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8749005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5558 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 11:37:16 +00:00
f92aaff104
AudioProcessing is not a Module.
...
Remove Module as the base class of AudioProcessing. The inherited
methods were all no-ops.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/8779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 04:22:49 +00:00
e2fc13e42f
Refactoring common_audio/signal_processing: Removed two macros used by isac only.
...
Removed a macro for malloc() and one for free(). They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.
BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org , turajs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:12:34 +00:00
38bf249049
Initialize output_will_be_muted_.
...
TBR=aluebs
Review URL: https://webrtc-codereview.appspot.com/8659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 17:43:44 +00:00
fcfc6a990e
Small refactoring of NetEq unittest for CNG with clock drift
...
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.
This change is only made for NetEq4.
R=tlegrand@google.com
Review URL: https://webrtc-codereview.appspot.com/8599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:42:28 +00:00
17342e5092
Add a method to inform AudioProcessing that its output will be muted.
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 22:28:31 +00:00
de782180b0
Change the type of propagation delta from int64 to int.
...
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.
BUG=2910
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 19:19:23 +00:00
07b5950c12
Initialize key_pressed_.
...
Was resulting in an error on Mac Asan:
[ RUN ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():
TBR=aluebs
Review URL: https://webrtc-codereview.appspot.com/8539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 16:41:13 +00:00
ce8e077cf0
Add a keypress field to the audioproc debug proto.
...
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.
TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.
R=aluebs@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 15:28:30 +00:00
75dd2885c5
Add an interface for accepting keypress signals to AudioProcessing.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 20:52:30 +00:00
8685af7ea0
Remove "Too long processing time of Incoming frame" logspam.
...
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.
BUG=2732
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 17:48:11 +00:00
a80be4b23c
Add boundary checking to supress gcc 4.8.3 warning.
...
BUG=2888
Test=try, voe_cmd_test
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 16:38:45 +00:00
82ebb463fd
Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
...
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.
It make the simliar feature's implementation consistent.
R=andrew@webrtc.org , fischman@webrtc.org , fischman@chromium.org
BUG=334447
Committed: https://code.google.com/p/webrtc/source/detail?r=5517
Review URL: https://webrtc-codereview.appspot.com/7769006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 04:48:27 +00:00
a65abf9d3a
Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file."
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This reverts commit 7686f0ddda717a9e776be0e219f039f68a10f9ed.
BUG=
TBR=andrew@webrtc.org , fischman@webrtc.org ,
Review URL: https://webrtc-codereview.appspot.com/8369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:26:26 +00:00
1f64f06784
Add stats of incoming frame delays for debugging bandwidth estimation.
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BUG=crbug/338380
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:12:14 +00:00
7686f0ddda
Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
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This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.
It make the simliar feature's implementation consistent.
R=andrew@webrtc.org , fischman@webrtc.org , fischman@chromium.org
BUG=334447
Review URL: https://webrtc-codereview.appspot.com/7769006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 17:42:34 +00:00
ad3035fc9e
Fix WindowCapturerWin to unselect bitmap before destroying DC.
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BUG=https://code.google.com/p/webrtc/issues/detail?id=2901
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 21:24:04 +00:00
