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1309 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
ce8e0936d9 Rename AutoMute to SuspendBelowMinBitrate 
Changes all instances throughout the WebRTC stack.

BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 12:18:43 +00:00
stefan@webrtc.org
b082ade3db Hook up audio/video sync to Call. 
Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:45:11 +00:00
henrik.lundin@webrtc.org
6e95d7afab Increment RTP timestamps for padding packets 
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.

A test was implemented to verify that the padding packets do
get their own timestamps.

BUG=2611
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-15 08:59:19 +00:00
stefan@webrtc.org
9b82f5a6ed Fix for RTX in combination with pacing. 
Retransmissions didn't get sent over RTX when pacing was enabled since
the pacer didn't keep track of whether a packet was a retransmit or not.

BUG=1811
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 15:29:21 +00:00
henrik.lundin@webrtc.org
e8433eb115 Reimplementing NetEq4's AudioVector 
The current implementation using std::vector is too slow.
This CL introduces a new implementation, using a regular
array as data container.

In AudioMultiVector::ReadInterleavedFromIndex, a special case for
1 channel was implemented, to further reduce runtime. Finally,
AudioMultiVector::Channels was reimplemented.

The changes in this CL reduces the runtime of neteq4_speed_test
by 33%.

BUG=1363
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 13:15:02 +00:00
asapersson@webrtc.org
38599510df Parse next RTCP XR report block after an unsupported block type. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 08:08:26 +00:00
minyue@webrtc.org
3e427263ee Reducing opus_test runtime to pass Android test 
BUG=2609
R=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 22:03:52 +00:00
andrew@webrtc.org
e03cafaebc MIPS optimizations for AECM audio processing module 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2279005

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 20:10:01 +00:00
andrew@webrtc.org
b0730108a2 Move audio_processing dependencies to a variable. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 17:20:27 +00:00
pbos@webrtc.org
57eb858698 Remove ".." from include_dirs in build/common. 
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 10:20:27 +00:00
andrew@webrtc.org
6e908b3adf Remove unnecessary include_dirs from audio_processing. 
TBR=aluebs
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/3659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 19:52:05 +00:00
stefan@webrtc.org
48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state. 
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.

Also makes sure that only valid timestamps and receive times are used for audio/video sync.

BUG=2608
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 15:18:52 +00:00
marpan@webrtc.org
bde3056567 Fix for video_processor_intergration_tests to run in parallel. 
BUG=2601.
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 20:59:29 +00:00
kjellander@webrtc.org
7a36cb408b Add missing dependencies to .isolate files 
Also fix invalid paths in video_engine_tests.isolate.

TEST=trybots passing compile step (no .isolate use is deployed on them yet)
BUG=chromium:300017
R=pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 14:28:57 +00:00
fischman@webrtc.org
b8cb85b348 Fix broken build on x86 Android 
BUG=2545
R=fischman@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3019004

Patch from Lu Quiang <qiang.lu@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 19:06:08 +00:00
asapersson@webrtc.org
766154aa1d Removed unused code. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 08:35:50 +00:00
sheu@chromium.org
5dd2ecb32d Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.

TBR=niklas.emblom@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/3269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 23:41:04 +00:00
sheu@chromium.org
74e6e8458e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:48:16 +00:00
sheu@chromium.org
d705649edf Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.

Revert while build breakage is fixed.

BUG=None
TBR=niklas.emblom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:20:15 +00:00
sheu@chromium.org
1a4ed0d70c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 20:32:28 +00:00
turaj@webrtc.org
58cd31665c Address Clag Analyzer issues. 
Following are the issues related to NetEq 4, discovered by Clang Analyzer.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath
Valid; perhaps unlikely, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath
Valid, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath
Valid; Addressed

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html
Not valid;

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath
Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath
Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath
Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath
Valid; addressed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:15:55 +00:00
asapersson@webrtc.org
7d6bd22019 Propagate estimated RTT from receivers to rtt observer. 
BUG=1613
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 12:14:34 +00:00
sergeyu@chromium.org
773e72797f Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc 
Chromium issue:
https://code.google.com/p/chromium/issues/detail?id=310146

BUG=2551
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2759004

Patch from Daniel Nicoara <dnicoara@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 01:51:21 +00:00
solenberg@webrtc.org
dce70ccb0b Add delay limit to ChokeFilter. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 19:18:07 +00:00
solenberg@webrtc.org
d6e46638ec Logging for BWE test framework. 
BUG=
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 16:06:26 +00:00
turaj@webrtc.org
55e1723713 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. 
BUG=2515
TEST=reproduced locally on linux and verified the fix resolves the issue.
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 04:40:09 +00:00
mikhal@webrtc.org
0aeb22e32c Adding tl0idx consideration for continuity 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 22:26:14 +00:00
henrik.lundin@webrtc.org
1a3a6e5340 Removing the threshold from the auto-mute APIs 
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.

BUG=2436
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 10:16:14 +00:00
sprang@webrtc.org
fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. 
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.

BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 09:21:07 +00:00
xians@webrtc.org
c94abd313e Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 18:15:09 +00:00
xians@webrtc.org
0729460acb Added a "interleaved_" flag to webrtc::AudioFrame. 
And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.

BUG=
TEST=compile
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 12:50:46 +00:00
henrik.lundin@webrtc.org
b56d0e383e Change the low-bitrate handling in BitrateControllerImpl 
Changing to using strategy classes rather than having two different
derived classes of BitrateControllerImpl. This enables run-time switching
of the strategy, which is now possible through a new API. The reason is
that it must fit the current design of ViE.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 09:24:06 +00:00
fischman@webrtc.org
37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. 
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 23:59:45 +00:00
andrew@webrtc.org
22858d4785 Add an extended filter option to audioproc. 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 14:07:17 +00:00
asapersson@webrtc.org
042e91c2b2 Fix for incorrect RTT estimation. A too low RTT value could be estimated. 
R=andrew@webrtc.org, holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 13:58:31 +00:00
andrew@webrtc.org
31628aae7e Upgrade scoped_ptr to Chromium's latest version. 
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.

- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.

TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 12:50:00 +00:00
andrew@webrtc.org
621df678c8 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. 
Mostly to remove a long-standing TODO...

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 10:27:23 +00:00
marpan@webrtc.org
943e3b95a6 Add CurrentLayerId() to temporal layers. 
same patch as: https://webrtc-codereview.appspot.com/2427004/

TBR=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/2729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 01:55:07 +00:00
solenberg@webrtc.org
8215106371 Framework for testing bandwidth estimation. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 14:23:26 +00:00
henrik.lundin@webrtc.org
29dd0de5b3 Changing the bitrate clamping in BitrateControllerImpl 
This CL implements an alternative to the bitrate clamping that is done
in BitrateControllerImpl. The default behavior is unchanged, but if
the new algorithm is enabled the behavior is as follows:
When the new bitrate is lower than the sum of min bitrates, the
algorithm will give each observer up to its min bitrate, one
observer at a time, until the bitrate budget is depleted. Thus,
with this change, some observers may get less than their min bitrate,
or even zero.

Unit tests are implemented.

Also fixing two old lint warnings in the affected files.

This change is related to the auto-muter feature.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 14:00:01 +00:00
pbos@webrtc.org
e05362916c Make sure the first frame isn't dropped. 
If frames were delivered within the same millisecond as VideoCaptureImpl
was created, or the timestamp weren't granular enough then the first
frame would be mistakenly dropped because of having the same timestamp
as a previous one, even though there was no previous one.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 09:02:30 +00:00
andrew@webrtc.org
89b1e688ca Minor comment fix after clang reformat. 
TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-18 14:23:29 +00:00
sergeyu@chromium.org
2df89c0c8b MouseCursorMonitor implementation for OSX and Windows. 
BUG=crbug.com/173265
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2388004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-17 19:47:18 +00:00
wu@webrtc.org
d030972139 Remove unused kPowTableFrac which causes anroid clang build failure. 
http://build.chromium.org/p/tryserver.chromium/builders/android_clang_dbg/builds/84322/steps/compile/logs/stdio

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2417004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 20:32:09 +00:00
sergeyu@chromium.org
e6e749da38 Add MouseCursorRenderer. 
The new class acts as a wrapper for DesktopCapturer interface. It takes
mouse shape and position from MouseCursorCapturer and renders it on the
frames produced by underlying DesktopCapturer.

BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)

Review URL: https://webrtc-codereview.appspot.com/2387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 02:48:41 +00:00
sergeyu@chromium.org
2767b53f66 Add MouseCursorCapturer interface with implementation for X11. 
The new interface will be used to capture cursor shape and position and
blend it into the image captured with desktop capturers.

BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)

Review URL: https://webrtc-codereview.appspot.com/2386005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 02:42:38 +00:00
kjellander@webrtc.org
3555303cb0 Roll chromium_revision 226126:228675 and fix clang warnings 
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.

This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.

TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 20:10:17 +00:00
stefan@webrtc.org
e5021fe590 Make RtpData and RtpFeedback destructors public. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 10:38:40 +00:00
andrew@webrtc.org
c2e471d8b3 Compile out unused kMinTrustedDelayMs. 
TBR=niklas.enbom@webrtc.org
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/2398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 02:11:21 +00:00
henrik.lundin@webrtc.org
1871dd2fb7 NetEq4: Removing templatization for AudioVector 
This is the last CL for removing templates in Audio(Multi)Vector.

BUG=1363
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2341004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 20:33:25 +00:00