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Author SHA1 Message Date
stefan@webrtc.org
ab800f64bc Disable flaky libjingle tests under tsan and memcheck. 
BUG=2380, 2379
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2218004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-16 15:22:32 +00:00
pbos@webrtc.org
5860de02aa Implement NACK over RTX for VideoSendStream. 
BUG=2231
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2197008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-16 13:01:47 +00:00
andresp@webrtc.org
8fa436bd65 Remove use of vcm->ResetDecoder from modules/utility. 
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2203006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-16 11:26:35 +00:00
phoglund@webrtc.org
62b816afcf Fixed pylint warnings. 
Passing variables into the page template with vars() is how it's regularly done AFAIK, so I'll just disable the warnings.

BUG=2371
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-16 11:26:12 +00:00
andrew@webrtc.org
15b8871e4a Allocate float_buffer_ in the initializer list. 
This may fix a Dr. Memory error: "allocated with operator new, freed
with operator delete[]". I suspect this is a false positive; in the
existing implementation the reset causes a delete[] on NULL. This is
a no-op of course, but Dr. Memory might be flagging it. We shall see.

In any case, this change is an improvement.

BUG=2321
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/2215004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-14 01:57:55 +00:00
sergeyu@chromium.org
8a1448950c Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams 
BUG=2374
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-14 00:30:51 +00:00
andresp@webrtc.org
f7eb75be1a Split VideoCodingModuleImpl into VideoSender and VideoReceiver. 
Only implmentation is changed the interface to the module is unchanged for now.

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-14 00:25:28 +00:00
sergeyu@chromium.org
a59696b2a5 Update libjingle to 52300956 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2213004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 23:48:58 +00:00
turaj@webrtc.org
48af652ea5 Prepare to compile ACM1 and ACM2. 
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 23:06:59 +00:00
wu@webrtc.org
bc189fb3b9 * Prefer to send ISAC on clank. 
* Add url option asc and arc to allow setting preferred audio send/receive codec.

TESTED=mobile as caller and callee:
pc-n7: pc sends opus, n7 sends isac 
pc-n4: pc sends opus, n4 sends isac
pc-pc opus-opus

R=braveyao@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2196006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 20:11:47 +00:00
sergeyu@chromium.org
6ab45b9dab Implement DesktopRegion subtraction. 
Region subtraction is used in chromoting client, so it's needed to
replace SkRegion.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2205004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 19:53:16 +00:00
andresp@webrtc.org
1f09dbe353 Moving test-only code (stream_generator) out of vcm implemention. 
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 19:17:54 +00:00
andrew@webrtc.org
2553450959 Fix win trybot errors due to r4729. 
The addition of logging.h in r4729 was causing the win trybot to fail
with "#pragma deprecated" errors in standard library headers. This
turned out to be due to including strsafe.h (via audio_device_config.h)
before sstream (via logging.h).

strsafe.h was only being included for the unused DEBUG_PRINT macro. I
removed all references to it.

This incidentally removes a bunch of other unneeded headers discovered
while trying to track the problem down.

This didn't show up in the commitbots; my guess is that the trybots are
using the VC10 toolchain and the commitbots the VC11 toolchain.

TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/2204004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 00:02:13 +00:00
sergeyu@chromium.org
6a5cc9d899 Fix crash in the window capturer on windows 
BUG=crbug.com/289753
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2203005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 19:17:26 +00:00
turaj@webrtc.org
7959e16cc2 ACM2 integration with NetEq 4. 
nack{.cc, .h, _unittest.cc} are basically copies from main/source/ folder, with cpplint warning cleaned up.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 18:30:26 +00:00
mallinath@webrtc.org
82a846f0cb Adding Ami to the video renderer and capturer modules. 
TBR=fischman@webrtc.org,wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2202006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:43:17 +00:00
fischman@webrtc.org
36cf4d2309 The video render module for iOS. 
BUG=2105, 2028
R=fischman@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2064004

Patch from SeungJae Lee <sjlee@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:39:53 +00:00
minyue@webrtc.org
e509f943ed This issue is related to 
https://chromereviews.googleplex.com/9908014/

I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.

BUG=
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:03:00 +00:00
andrew@webrtc.org
8fa03a15ab Make PCM16 available in Chromium builds. 
PCM16 can be useful for unit tests in Chromium. In particular Mikhal
would like to use it for ChromeCast.

This currently (r222592) has no impact on Chrome binary size, presumably
because PCM16 is unused and the linker strips the symbols.

To measure the potential impact, I looked at the size (bytes) of
out/Release/vie_auto_test on Linux with various codecs removed:
r4724    : 4567384
No PCM16 : 4565936
No ILBC  : 4500424
No G722  : 4555800
No RED   : 4565880

Giving the following size increases of adding each codec:
PCM16 :  1.4 kB (0.03%)
ILBC  : 70.0 kB (1.49%)
G722  : 11.6 kB (0.25%)
RED   :  1.5 kB (0.03%)

R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2195005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:30:30 +00:00
andrew@webrtc.org
89df092807 Make the destructor of AudioCodingModule public. 
This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:27:43 +00:00
andrew@webrtc.org
5eb997a2fd Fix unsigned/signed comparison error due to r4729. 
Fix it for real by switching to ints rather than casting.

TBR=xians

Review URL: https://webrtc-codereview.appspot.com/2191009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:01:42 +00:00
andrew@webrtc.org
8f94013651 Reduce frequency of high audio delay warning logs. 
This will log the warning every 5 seconds instead of every 10 ms.

BUG=b/10674993
TESTED=Ran voe_cmd_test with hard-coded high delay. Observed a log
every 5 seconds.

R=noahric@chromium.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 22:35:00 +00:00
henrike@webrtc.org
256b83146c Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. 
BUG=2364
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 20:43:13 +00:00
pbos@webrtc.org
5c678eabd9 Implement 'abs-send-time' extension in VideoSendStream. 
BUG=2229
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 19:00:39 +00:00
henrike@webrtc.org
6138c5cfa4 OpenSl: fixes crashes externally reported in issue 2361 and 2362. 
BUG=2361,2362
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 18:50:06 +00:00
turaj@webrtc.org
036b7436df Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. 
Un-implemented APIs.

TBR=henrik.lundin@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2191008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 18:45:02 +00:00
braveyao@webrtc.org
a80ee74f69 AppRTC: using a footer element instead of div#footer in CSS. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2200004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 16:24:07 +00:00
mikhal@webrtc.org
d4d59ac871 Remove FrameForStorage:Follow up on r4688 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2201004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 15:18:15 +00:00
pbos@webrtc.org
2902328cce Implement 'toffset' extension in VideoSendStream. 
BUG=2229
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 10:14:56 +00:00
stefan@webrtc.org
554d158ce6 Reset jitter buffer and timing if frames are getting too much delay. 
BUG=chromium/263867
TEST=trybots
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 08:45:26 +00:00
andrew@webrtc.org
835ef67d14 Remove repeated conditions key. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 00:16:00 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android. 
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 18:24:07 +00:00
braveyao@webrtc.org
641340944b Show the signaling state and ice connection state in AppRTC by hooking up the peerconnections .onsignalingstatechange and .oniceconnectionstatechange events. 
Hopefully this will increase the quality of the "it does not work" reports from users by giving them more information about what is going on under the hood.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2174004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4718 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 17:37:16 +00:00
pbos@webrtc.org
319c98d663 Fix format string in video_quality_analysis.cc. 
Fixes compilation errors on Android and Linux32 targets.

TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2196005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 15:23:50 +00:00
pbos@webrtc.org
182d025d94 Remove include_dirs from voice_engine.gyp. 
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2193004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 15:01:26 +00:00
pbos@webrtc.org
df531a2eee Test that VideoSendStream responds to NACK. 
BUG=2228
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 14:56:33 +00:00
kjellander@webrtc.org
f880f863dd Convert printing in video quality tests to Chromium's perf format. 
Add support for --label flag to the frame_analyzer, that
decides what label shall be used for the perf output.

BUG=none
TEST=
Make sure to have zxing and ffmpeg in the PATH.
Create a captured video (from running vie_auto_test custom call)
webrtc/tools/compare_videos.py --ref_video=reference_video.yuv --test_video=captured_output.yuv --frame_analyzer=out/Release/frame_analyzer --label=TEST_VGA
And then inspecting the output that is prefixed with RESULT.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 12:10:01 +00:00
pbos@webrtc.org
e07049f19f Lock RTPSender statistics. 
Suppressing these errors in TSan has become tedious. It's better to just
lock them.

BUG=2349
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2197004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 11:29:17 +00:00
pbos@webrtc.org
744fbc7fe4 Split up EngineTests and RampupTests. 
This allows having one group of tests per file, the test files are
long enough as they are.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 09:26:25 +00:00
andrew@webrtc.org
eda189be14 Remove redundant STR_CASE_CMP macro definitions. 
R=minyue@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2187005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:50:10 +00:00
elham@webrtc.org
a19c9f4173 Updated WebRTC version to 3.41 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:23:44 +00:00
pbos@webrtc.org
021c42bfa8 Lock use of _packetRequestCallback in VCM. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:18:31 +00:00
pbos@webrtc.org
7ebf0e7f44 Remove include_dirs from video_engine_core.gypi. 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 16:56:31 +00:00
pbos@webrtc.org
59f20bb735 Break out RTCPSender dependency on ModuleRtpRtcpImpl. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 16:02:19 +00:00
pbos@webrtc.org
26b0d77baf Suppress RTPSender race regardless of codec. 
New test uses SendGeneric instead of SendVP8.

BUG=2349
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 15:34:36 +00:00
pbos@webrtc.org
841c8a44bb Rename VideoCall to Call. 
Call should encompass more than video, there's no point in calling it
VideoCall.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2191005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 15:04:25 +00:00
solenberg@webrtc.org
86136a0e8f Re-enable tests for Remote Bitrate Estimator 
BUG=
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 13:06:52 +00:00
pbos@webrtc.org
0181b5f8dd ExternalVideoDecoder for new VideoEngine API. 
Implements the ExternalVideoDecoder interface for VideoReceiveStream.
Also adds a FakeDecoder used in tests, removing the overhead of running
the EngineTest tests with VP8 under Memcheck/TSan, allowing us to enable
them under Memcheck/TSan as well.

BUG=2346,2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2172004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 08:26:30 +00:00
pbos@webrtc.org
30e055c4dd Handle empty RTP video packets agnostic to codec. 
Sending empty RTP packets caused a crash when using a generic codec
instead of VP8. This fix moves handling of empty RTP packets out of
ReceiveVp8Codec and into ParseRtpPacket.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2185004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-08 11:15:00 +00:00
mallinath@webrtc.org
1b476d9a56 Disabling channelmanager unittest. This test is causing 
TSAN error. The problem could be in thread Invoke method.

TBR=wu@webrtc.org
BUG=https://code.google.com/p/webrtc/issues/detail?id=2355

Review URL: https://webrtc-codereview.appspot.com/2190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-07 18:59:12 +00:00