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Author SHA1 Message Date
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
elham@webrtc.org
9c735c4e25 Updated WebRTC version to 3.45 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5009 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 16:34:50 +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
henrik.lundin@webrtc.org
0d19ed9a06 AutoMute: Adding channel_id parameter to callback. 
BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 12:37:13 +00:00
pbos@webrtc.org
fe1ef935e7 Implement I420FrameCallbacks in Call. 
BUG=2425
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2393004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 10:34:43 +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
kjellander@webrtc.org
eb61a851d5 Move audio_e2e_harness into include_tests==1 condition. 
To avoid compile errors when WebRTC is built as a part of
Chromium.

TEST=ran gclient runhooks locally.
BUG=none
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5003 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 08:40:56 +00:00
kjellander@webrtc.org
88a310886e Add audio_e2e_test target to tools.gyp 
The moving this GYP target out of webrtc.gyp in
https://code.google.com/p/webrtc/source/detail?r=4949
this should have been added into tools.gyp.

TEST=trybots passing
BUG=none
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-19 18:10:04 +00:00
wu@webrtc.org
fb648da2b9 Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. 
BUG=2508
RISK=P1
TEST=try bots
R=henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2425004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-18 21:10:51 +00:00
stefan@webrtc.org
3e00505e9a Have padding decay to zero if no frames are being captured. 
BUG=1837
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-18 15:05:29 +00:00
kjellander@webrtc.org
893c07f209 Disable the -Wno-unused-const-variable Clang warning on Mac 
In r4966 this was disabled on Linux for WebRTC code in order to detect any new unused const variables.
This CL does the same for Mac.

BUG=none
TEST=added an unused const and verified compilation fails
when this patch is applied. Mac trybots passing as well.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-18 14:42:27 +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
6342066974 Fix tsan failures in channel.cc regarding to the volume settings. 
BUG=2461
TEST=try bots
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-17 18:28:55 +00:00
xians@webrtc.org
675e260ad1 Check the number of playout channels instead of the send channels in StopPlayout() 
BUG=2467
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2420004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-17 16:15:34 +00:00
pbos@webrtc.org
c11148b352 Compound/reduced-size RTCP in VideoReceiveStream. 
BUG=2424
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2413004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-17 14:14:42 +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
sprang@webrtc.org
25fce9adc5 Fixed issue with how MTU is calculated. 
BUG=
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2410004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 13:29:14 +00:00
stefan@webrtc.org
b400aa7cd4 Don't pad if only one stream is sent, except if auto muted. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2406004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 13:03:10 +00:00
kjellander@webrtc.org
e7009f3b0d Revert "Disable tests for TSan v2" 
This reverts r4787 since r4966 got us a newer Clang
version with these issues fixed.

BUG=2259,2334
TEST=test_support_unittests and system_wrappers_unittests passed execution under TSan v2 for 100 iterations:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release test_support_unittests system_wrappers_unittests

TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/system_wrappers_unittests --gtest_repeat=100 --gtest_break_on_failure

TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/test_support_unittests --gtest_repeat=100 --gtest_break_on_failure
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2407004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 12:26:48 +00:00
sprang@webrtc.org
5d957e29f7 Wired up max packet size and added simple test. 
BUG=2428
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 11:37:54 +00:00
pbos@webrtc.org
9401524211 Run FullStack tests without render windows. 
Also disables test on valgrind platforms, it has no chance to keep up.

BUG=2278
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2159008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 11:05:37 +00:00
kjellander@webrtc.org
5ed4f46af1 Remove TSan v2 disabled test in condition_variable_unittest.cc 
When we rolled our chromium_revision 226126:228675 in r4966
we picked up Clang r191856, which fixes the problem we've
seen earlier in condition_variable_unittest.cc.
Because of this, I'm now re-enabling this test.

TEST=trybots passing
BUG=2259
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2404004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 10:50:41 +00:00
pbos@webrtc.org
b44c2a3193 Open file in binary in CreateFromYuvFile(). 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-16 08:46:06 +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
pbos@webrtc.org
266c7b330a Move ChromaGenerator to common_video/. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 09:15:47 +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
henrike@webrtc.org
901ae77618 Android: Fixes WebRTCDemo build (missing Java code). 
TBR=ajm@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2395005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 21:46:53 +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
sergeyu@chromium.org
30792987b8 Remove empty line in SharedXDisplay::RemoveEventHandler. 
TBR=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2397004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 17:58:46 +00:00
henrike@webrtc.org
05773e5a70 Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. 
TBR=fischman@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2395004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 16:25:11 +00:00
sergeyu@chromium.org
7419a72383 Add event handling in SharedXDisplay. 
SharedXDisplay has to handle X events because the events may belong to
different clients of that class.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2386004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-13 00:44:09 +00:00
sergeyu@chromium.org
894e6fe9ea Add DesktopCaptureOptions class. 
The new class is used to pass configuration parameters to screen/window
capturers. It also allows to share X Window connection between multiple
objects.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2374004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-12 22:40:05 +00:00
henrike@webrtc.org
f53622d42e WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. 
BUG=2083
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-11 21:28:26 +00:00
kjellander@webrtc.org
5b3b6b1784 Reorganize GYP targets to make webrtc.gyp more usable. 
When WebRTC is built as a part of Chromium, some of
the stuff in webrtc.gyp will not be found. This CL
fixes this.

TEST=trybots passing. I also did some manual builds for Android with the android_builder_webrtc target in build/all_android.gyp of a Chromium checkout.
BUG=chromium:304143
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2353004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-10 08:48:16 +00:00
andrew@webrtc.org
13b2d46593 clang-format audio_processing/aec/* 
TBR=bjornv
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/2373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-08 23:41:42 +00:00
andrew@webrtc.org
ca764ab22d Add a parameter to audioproc for overriding the delay. 
Rename the parameter for adding to the input delay to "add_delay".

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2345007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 16:44:32 +00:00
elham@webrtc.org
11e9cbc399 Updated WebRTC version to 3.44 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2365004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 16:18:35 +00:00
stefan@webrtc.org
f5d7c5891c Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." 
Revert r4935 "Fix build error in r4934."

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2364004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:42:46 +00:00
stefan@webrtc.org
611e5141cb Fix build error in r4934. 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2363004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:35:36 +00:00
stefan@webrtc.org
bc99bcfa6f Add a tool for parsing an RTP file and outputting the BWE relevant fields. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2237005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:21:24 +00:00
turaj@webrtc.org
6d5d248075 Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident. 
BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-06 04:47:28 +00:00
turaj@webrtc.org
f31639612d Accounting for wrap-around of timestamps. 
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-06 02:21:24 +00:00
mikhal@webrtc.org
35e4dd3067 VPM: Fixing namespace 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4930 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 20:21:30 +00:00
fischman@webrtc.org
4598380860 Android: enable camera video stabilization when available. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2347005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 20:14:19 +00:00
kjellander@webrtc.org
7fca2ce097 Add owners to [webrtc,talk]/build and *.isolate (take 2) 
After fischman@'s comments in http://review.webrtc.org/2347006/ here's another CL to clean up the redundancies and add wu@ to webrtc/build/

TEST=none
BUG=none
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2348006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 19:36:45 +00:00