a9890800e0
Update talk to 58127566 together with
...
https://webrtc-codereview.appspot.com/5309005/ .
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:21:03 +00:00
a92baead39
ACM 2 compatibility with ACM 1.
...
Removing an unregisterd codec from ACM 1 does not result in an error, so should be for ACM 2. Also ACM 1 has post-decode VAD on and AMC 2 needs to have it on by default.
BUG=
Test=trybits
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5276 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:10:44 +00:00
2018269dc3
Revert 5274 "Update talk to 58113193 together with https://webrt ..."
...
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
>
> R=mallinath@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:54:25 +00:00
a129b6cd13
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
...
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:40:39 +00:00
451745ec05
Complete rewrite of demo application.
...
BUG=2122
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 16:55:37 +00:00
88ac63abc6
Remove overloaded CpuOveruseMeasure function.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5272 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 14:37:33 +00:00
9ee75e9c77
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
...
BUG=N/A
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:42:44 +00:00
391b4db7de
Fix common_video_unittests in apk_tests.gyp.
...
r5265 moved common_video_unittests to its own gyp, this required an
update of apk_tests.gyp that wasn't caught by our trybots.
TBR=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5266 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:48:53 +00:00
724947b8ef
Add SwapFrame() to VideoSendStreamInput.
...
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +00:00
096e8d9f94
Revert 5259 "Callback for send bitrate estimates"
...
CL is causing flakiness in RampUpTest.WithoutPacing.
> Callback for send bitrate estimates
>
> BUG=2235
> R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4459004
R=mflodman@webrtc.org , pbos@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/5579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 14:07:33 +00:00
f9bdbe3619
Roll chromium_revision 232627:238260
...
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
2656cf9f4c
Callback for send bitrate estimates
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 12:53:03 +00:00
7ae8495779
Removed unnecessary Pulse init from VoE startup.
...
Saves 10% (~260ms) of the total PeerConnectionTest wallclock time.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5254 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 21:01:34 +00:00
762fcdcca9
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
...
g++ 4.7 and later support explicit virtual overrides when building with C++11
support enabled. However, webrtc does not detect that and makes OVERRIDE a
no-op.
This CL updates typedefs.h to define OVERRIDE properly when g++ 4.7 is used
with C++11 support enabled.
See this page for GCC support of C++11 features:
http://gcc.gnu.org/projects/cxx0x.html
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5149005
Patch from Chris Dumez <ch.dumez@samsung.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5253 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 19:20:46 +00:00
8b8819262f
Improve VideoSendStreamTest::MaxPacketSize
...
This CL was submitted as issue https://webrtc-codereview.appspot.com/4849004/ , but was reverted because of flakiness. This new issue will correct that.
Patch Set 1 contains the code that was submitted in 4849004.
BUG=2428
R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5251 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 10:05:17 +00:00
917306d3fd
Change uses of the obsolete armv7 setting to arm_version==7.
...
BUG=http://crbug.com/234135
R=andrew@webrtc.org , fischman@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5369004
Patch from Mostyn Bramley-Moore <mostynb@opera.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 09:26:07 +00:00
eb7def234e
Fix compilation errors on Fedora 20.
...
peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.
BUG=2700
R=andrew@webrtc.org , fischman@webrtc.org , henrik.lundin@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5019004
Patch from Victor Costan <costan@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 21:34:30 +00:00
de7c9e8884
Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.
...
Move the logic to common.gypi to reduce the chance of the define being
unprotected in the future.
BUG=b/12018143
TESTED=git try, and local Linux build with -Denable_video=0
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5244 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 16:23:00 +00:00
5e13ac967b
Add shape in DesktopFrame.
...
The shape will be used for Me2App mode in chromoting.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5243 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-07 01:03:28 +00:00
8d0ca7f5d2
Add new method to MockAudioProcessing.
...
TBR=henrikg
Review URL: https://webrtc-codereview.appspot.com/5279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5241 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 17:52:27 +00:00
797522f9f2
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
...
> Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
>
> BUG=2428
> R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4849004
It caused a failure in video_engine_tests on the Linux Tsan bot.
TBR=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5240 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 17:42:32 +00:00
863b536100
Allow opening an AEC dump from an existing file handle.
...
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.
This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.
BUG=2567
R=andrew@webrtc.org , henrika@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 16:05:17 +00:00
0f3d0bb601
Stop video capturers in multi-stream test.
...
Expected to reduce runtime and flakiness in
CallTest.SendsAndReceivesMultipleStreams on linux_memcheck which is
presumed to be due to contention between the threads.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5238 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 15:48:17 +00:00
88615f0659
Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5236 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 13:16:44 +00:00
7f73280ded
Fraction lost statistics not being reported
...
A bug is causing fraction lost to always be set to zero when calling
ViERTP_RTCP::Get(Send|Receive)ChannelRtcpStatistics. Fix this and update
tests to catch it.
BUG=
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5235 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 11:56:55 +00:00
7104fc1906
Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
...
BUG=2428
R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5229 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 16:15:11 +00:00
96a9b2dcdc
Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
...
R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/5049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 15:06:56 +00:00
ebad765ee0
Add callbacks for send channel rtp statistics
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:29:02 +00:00
5cea89f3e1
Remove CallTest dependency on voice_engine/test/.
...
Loading file out of resources/ instead of data/ which is deprecated.
BUG=
R=holmer@google.com , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5226 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:24:17 +00:00
0a3c1471b8
Add API to query video engine for the send-side delay.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:05:07 +00:00
07fcc4f2fa
Fixing the android build
...
The build broke due to r5222.
BUG=2436
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5224 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 13:24:25 +00:00
c49d5b7df8
Move implementation files out of the webrtc/ root.
...
Leaves the root for public headers. Also fixes the issue of requiring
root OWNERS approval for changes in the Call implementation and adding
end-to-end tests.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 12:11:47 +00:00
245037df09
Remove default implementations for SuspendBelowMinBitrate
...
These two methods had default implementations while waiting for
changes in libjingle to propagate. Now the changes are in, and
the default implementations are removed.
BUG=2436
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5222 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 12:01:45 +00:00
b88fc18aba
Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5221 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 11:36:46 +00:00
a6ad6e5b58
Add callbacks for send channel rtcp statistics
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 09:48:44 +00:00
c4726d06fa
Make RTPSender::SendPadData public.
...
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5219 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 09:16:33 +00:00
3d9981d58a
Remove unused ThreadData struct.
...
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/4949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5216 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 17:13:47 +00:00
3054ba6bb2
Remove the long disabled WEBRTC_SVNREVISION define.
...
BUG=500
TESTED=git try
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5215 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 17:00:44 +00:00
5b51ebc179
Removing DropDeltaAfterKey functionality which is unused.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5214 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 15:53:24 +00:00
71f055fb41
Add send frame rate statistics callback
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 15:09:27 +00:00
9e5b0342f6
Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5212 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 13:47:44 +00:00
79b63206b9
Fixes a crash in fullstack tests introduced with r5209.
...
TBR=mflodman@webrtc.org
BUG=1812
Review URL: https://webrtc-codereview.appspot.com/4689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5211 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 13:34:28 +00:00
b477fa6d21
Small fixes to plot_neteq_delay.m
...
Fixing problems with wrap-arounds and other small things. Adding an
extra output value.
Review URL: https://webrtc-codereview.appspot.com/4929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5210 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 12:28:47 +00:00
7e9315b42e
Adds support for sending redundant payloads over RTX.
...
TEST=trybots
BUG=1812
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 10:24:26 +00:00
9523b55826
Fix a typo in neteq.gypi
...
This CL is for NetEq3. The #define for iSAC-fb was wrong on one
line. It did not affect the defualt use case, but resulted in
errors if 48 kHz mode was enabled.
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5208 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 08:24:49 +00:00
d7696c4ed1
Compile-out functions only used by the bit-exact test.
...
Causes errors on platforms where the test is unused.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/4869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5207 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 23:39:16 +00:00
d3865e9124
Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close).
...
It is incorrect to wrap close in HANDLE_EINTR on Linux.
BUG=chromium:269623
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4759004
Patch from Mark Mentovai <mark@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5206 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 19:10:20 +00:00
812dd11f8c
Add baseline generation/verification to BWE test framework.
...
Updating resource file separately, once LGTM. Generates ~628k of files for current tests, highly compressable, once/if we need that.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5204 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 15:11:14 +00:00
499631c1e4
Utility class for reading/writing network-byte-ordered integers.
...
BUG=
R=holmer@google.com , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2151008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5203 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 13:22:48 +00:00
37968a9be7
Change BitrateStats to more generalized RateStatistics
...
BUG=2656
R=holmer@google.com , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5202 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 10:31:59 +00:00
