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Author SHA1 Message Date
andresp@webrtc.org
2397a17c6b Fix bug introduced during replace of list wrapper with std equivalents in r5378. 
R=henrika@webrtc.org, pbos@webrtc.org, henrike@webrtc.org
TBR=henrike@webrtc.org
BUG=2164

Review URL: https://webrtc-codereview.appspot.com/7639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 12:33:30 +00:00
minyue@webrtc.org
c8b99a49d1 This is to roll a more recent Chromium, which contains latest Clang, so as to be able to roll Opus 1.1, which will the next step. 
There are uninitializion problem with normal_asyn_test.cc. This is fairly easy to solve and therefore is included in this CL.

The following is a memo on the selection of the version to roll. It may be a reference for similar missions.

How was this version picked?

1. The whole purpose of this work is to update to Clang to be able to compile Opus 1.1. In Chromium, Clang got updated to 198389 at r244540.

2. From r245412, gyp_chromium requires "tools\find_depot_tools.py". However, WebRTC does not sync up the root of folder "tools". An issue has been created to Chromium on this.

... So the version must be a good version between r244540 and r245411 (inclusive)

BUG=

TEST=passes all trybots
R=kjellander@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 12:21:42 +00:00
sprang@webrtc.org
c00adbed73 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket 
StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
  be cached while holding lock to avoid race condition.

  Also, rtp_callback_ do not need to be called in GetStatistics() at all

BUG=2853
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 10:42:48 +00:00
pbos@webrtc.org
99eab02fb1 Fix "field '_testNo' is uninitialized" warnings. 
BUG=2849
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 09:30:35 +00:00
pbos@webrtc.org
c98882dcd3 Always initialize Trace in Call TraceDispatcher. 
Prevents violation of lock order occuring previously when
RegisterCallback called SetTraceCallback while holding its lock, which
called Print back (which acquires the lock).

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 09:11:10 +00:00
braveyao@webrtc.org
37c2976511 Samples, add IPv6 supporting into Apprtc demo. 
BUG=2828
TEST=Manual Test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/7509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 03:08:16 +00:00
andrew@webrtc.org
e84978f3d8 Add a Config parameter to AudioProcessing::Create(). 
Also add a parameter-less version; the (int) version is deprecated and
should be removed.

TBR=aluebs,bjornv
BUG=2844

Review URL: https://webrtc-codereview.appspot.com/7609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-25 02:09:06 +00:00
wu@webrtc.org
256d0ada35 Remove the check for audio codec num in WebRtcVoiceEngineTest.HasCorrectCodecs. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 18:58:51 +00:00
henrike@webrtc.org
57f6c10d00 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. 
BUG=2807(second issue)
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 18:42:12 +00:00
wu@webrtc.org
98aefcd8fe Update tsan suppressions for libjingle_media_unittest. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/7559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 18:32:43 +00:00
wu@webrtc.org
ca5ff9972e Re-enable webrtcvoice/videoengine unittests. 
TEST=try bots
BUG=
R=mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=5387

Review URL: https://webrtc-codereview.appspot.com/7149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 17:37:46 +00:00
asapersson@webrtc.org
871d949299 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 13:23:49 +00:00
andresp@webrtc.org
24999d44c2 Allow ?audio=false&video=false to be used in combination to instantiate a recv-only client. 
R=braveyao@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/6819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 12:25:50 +00:00
pbos@webrtc.org
fd0f267bb1 Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. 
Required to be able to link new API code against the merged target.
Replaces old dependency on video_engine_core as the new-API target
depends on it for now, and video_engine_core is being phased out.

R=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/7519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 10:43:47 +00:00
stefan@webrtc.org
99a8c7e039 Add trace-based delivery filter to BWE test framework. 
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 10:00:27 +00:00
pbos@webrtc.org
c279a5d72c Wire up RTX in VideoReceiveStream. 
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.

BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 09:30:53 +00:00
andresp@webrtc.org
8d375c95b7 Fix deadlock on register/unregister observer while there is a an going callback. 
BUG=2835
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 23:09:25 +00:00
wu@webrtc.org
a8910d2f88 Update talk to 60094938. 
Review URL: https://webrtc-codereview.appspot.com/7489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 22:12:45 +00:00
andrew@webrtc.org
754de528b7 Fix array declarations in aec_rdft.h. 
Was causing warnings in Chromium such as:
warning C4742: 'rdft_wk2i' has different alignment in
'webrtc\modules\audio_processing\aec\aec_rdft_sse2.c' and
'webrtc\modules\audio_processing\aec\aec_rdft.c': 4 and 16

BUG=chromium:336620
R=cduvivier@google.com

Review URL: https://webrtc-codereview.appspot.com/7489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 20:55:14 +00:00
pbos@webrtc.org
e7223e7795 Set NACKed packet to -1 in TestNackRetransmission. 
Zero is a valid sequence number which may occur even if there are no
retransmissions, this caused the test to flake as an incoming packet
would be mistaken for a retransmission.

BUG=2830
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 16:14:34 +00:00
sprang@webrtc.org
0e93257cee Add callbacks for receive channel RTP statistics 
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 10:00:39 +00:00
henrike@webrtc.org
91db93d24f Android, fixes crash on devices with only front cameras. 
BUG=2807
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 21:31:24 +00:00
kjellander@webrtc.org
570bc3d792 Make new baremetal trybots receive tryjobs by default. 
I've done several green builds with these machines, but I suspect
some of the flakiness we still see in the build waterfall may
occur on these ones. Hopefully at least the ones for vie_auto_test
will be ironed out in Q1 as the old Video Engine API becomes deprecated.

TEST=none
BUG=chromium:332726
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 19:00:01 +00:00
mallinath@webrtc.org
0d92ef67c4 Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation". 
This CL is attempting to correct those.

BUG=2810
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 02:21:22 +00:00
mallinath@webrtc.org
68cbd01216 enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable. 
BUG=2799
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 00:16:46 +00:00
andrew@webrtc.org
7de3bb9df9 Output logs to stderr from voe_cmd_test by default. 
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.

TESTED=logs output to stderr by default and to the usual file when the
flag is specified.

R=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 22:17:43 +00:00
henrike@webrtc.org
28da47c52f Android example apps: fixes issue where useful failure information was suppressed. 
BUG=2808
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 19:03:51 +00:00
fbarchard@google.com
1d2c034861 libyuv use extern c around jpeg includes. includes fixes to gyp build for intel/mips android, cros arm, ios, and pnacl. 
BUG=none
TESTED=try bots
R=andrew@webrtc.org, jzern@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7179005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 18:27:05 +00:00
sprang@webrtc.org
7dba27c740 Potential dead lock in receive statistics 
A dead lock could occur if the following to code paths are called
concurrently:

ReceiveStatisticsImpl::IncomingPacket() ->
  StreamStatisticianImpl::IncomingPacket()

StreamStatisticianImpl::GetStatistics() ->
  ReceiveStatisticsImpl::StatisticsUpdated()

Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().

BUG=2818
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:33:37 +00:00
elham@webrtc.org
32c3247418 Fix for libtalkmobile build error 
bug=b/12549061

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:16:58 +00:00
henrike@webrtc.org
7ef7df57d8 Removes script for generating supplement.gypi also adds git ignore for tools/gn. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 15:54:56 +00:00
pbos@webrtc.org
e02d47515f Set up receiver RTX config using a std::map. 
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 14:43:55 +00:00
asapersson@webrtc.org
efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 08:34:49 +00:00
henrike@webrtc.org
32c26eb90b Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback 
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-17 23:12:51 +00:00
jiayl@webrtc.org
4985927d36 Implement screen enumeration and individual screen capturing for Windows. 
BUG=2787
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-17 17:19:16 +00:00
henrike@webrtc.org
ead202b973 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. 
BUG=2801
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 23:26:37 +00:00
henrike@webrtc.org
2ce9a64b75 Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. 
BUG=12545067
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 16:49:53 +00:00
henrike@webrtc.org
0af1ffa84d Android, WebRTCDemo: fix issue where changing remote IP was not working properly. 
BUG=2783
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 16:45:15 +00:00
aluebs@webrtc.org
4ffd9c7423 Add full path to headers 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 09:01:39 +00:00
bjornv@webrtc.org
6a94734d4d Adds back set_sample_rate_hz() when Init is called in recordings. 
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.

BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 08:41:09 +00:00
andrew@webrtc.org
ea9392d5eb MIPS optimizations for NS audio processing module 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139006

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 07:22:01 +00:00
sergeyu@chromium.org
fb4e256d49 Fix crash in MouseCursor::CopyOf() 
This issue was causing test failures with the latest webrtc roll.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 04:45:35 +00:00
andrew@webrtc.org
8f35afab8c Exclude protoc objects from merge_libs.py. 
BUG=b/12567343
R=wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 00:31:57 +00:00
sergeyu@chromium.org
4b26e2eee3 Update libjingle to 59676287 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 23:15:54 +00:00
mallinath@webrtc.org
7a2ca7c621 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. 
This change is also must for rolling webrtc in chrome.

R=jiayl@webrtc.org
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 19:00:13 +00:00
wu@webrtc.org
8f19cb9fbc Revert 5387 "Re-enable webrtcvoice/videoengine unittests." 
Missed the result from the last try bot.

> Re-enable webrtcvoice/videoengine unittests.
> 
> TEST=try bots
> BUG=
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7149004

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 22:31:11 +00:00
wu@webrtc.org
eda6823397 Re-enable webrtcvoice/videoengine unittests. 
TEST=try bots
BUG=
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 22:15:09 +00:00
jiayl@webrtc.org
017b619010 Extends the ScreenCapturer interface for individual display screen cast. 
Real implementations for each platform will be added in future CLs.

BUG=2787
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/6819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 18:26:37 +00:00
wjia@webrtc.org
03cfde2d10 Roll Chromium 238260 -> 243863 
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:48:34 +00:00
andresp@webrtc.org
8c5b27de9a Allow to skip turn by passing ts=false to apprtc. 
R=braveyao@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:00:23 +00:00