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Author SHA1 Message Date
turaj@webrtc.org
167b6dfc73 Fix jitter buffer delay estimate. 
BUG=b/12099925
R=niklas.enbom@webrtc.org, niklase@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 21:05:07 +00:00
wu@webrtc.org
24301a67c6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 19:17:43 +00:00
mflodman@webrtc.org
92c2793154 Adding REMB to receive stream configuration, the send side will always 
react to incoming REMB for now.

Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.

TEST=See above.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 16:36:28 +00:00
asapersson@webrtc.org
86bb56a7f5 Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." 
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> 
> R=holmer@google.com
> 
> Review URL: https://webrtc-codereview.appspot.com/5049004

TBR=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 16:16:45 +00:00
pbos@webrtc.org
0a222eba69 Merge metrics_unittests into video_engine_tests. 
metrics_unittests will be removed as soon as trybots catch up with LKGR,
that way we don't have to break any tryjobs during.

BUG=1843
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 14:31:47 +00:00
pbos@webrtc.org
1d096901ac Move realtime tests to webrtc_perf_tests. 
New binary not to be run on our VMs as they result in flaky tests. These
will instead be run on baremetal machines.

BUG=2710
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 12:48:05 +00:00
sprang@webrtc.org
6811b6e308 Callback for send bitrate estimates - new roll 
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.

Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.

The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:

webrtc::RTPSender::BitrateUpdated()  // Get RTPSender stats lock
webrtc::Bitrate::Process()  // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...

webrtc::Bitrate::Update()  // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats()  // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...

This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.

BUG=2235
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 09:46:59 +00:00
mflodman@webrtc.org
f3973e81d5 Make sure channels in the same call are in the same channel group. 
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 09:40:45 +00:00
henrik.lundin@webrtc.org
e9abd591d7 Making RemoteRateControl::min_configured_bit_rate_ configurable 
The minimum bitrate can now be configured from WrappingBitrateEstimator.

BUG=2698
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 08:42:42 +00:00
wu@webrtc.org
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
turaj@webrtc.org
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
wu@webrtc.org
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
wu@webrtc.org
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
henrike@webrtc.org
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
asapersson@webrtc.org
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
henrike@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
sprang@webrtc.org
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
kjellander@webrtc.org
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
sprang@webrtc.org
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
fischman@webrtc.org
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
andrew@webrtc.org
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
sprang@webrtc.org
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
kjellander@webrtc.org
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
fischman@webrtc.org
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
andrew@webrtc.org
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
sergeyu@chromium.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
henrikg@webrtc.org
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
pbos@webrtc.org
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
sprang@webrtc.org
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
sprang@webrtc.org
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
sprang@webrtc.org
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
asapersson@webrtc.org
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
sprang@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
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
henrik.lundin@webrtc.org
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
pbos@webrtc.org
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
henrik.lundin@webrtc.org
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
stefan@webrtc.org
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
sprang@webrtc.org
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
stefan@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
andresp@webrtc.org
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
sprang@webrtc.org
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
asapersson@webrtc.org
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