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
hta@webrtc.org
758db4baea
Demo showing how to measure volume using WebAudio
...
This adds a page to the demos page, it does not affect any running code.
BUG=
R=dutton@google.com , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5237 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 14:47:34 +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
sergeyu@chromium.org
32f485b16a
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:36:21 +00:00
sergeyu@chromium.org
57a5f64264
revert r5230
...
r5230 broke windows build.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5232 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:14:46 +00:00
sergeyu@chromium.org
a1b21cd777
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5230 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 21:28:34 +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
sergeyu@chromium.org
5bc25c41fc
Update libjingle to 57692857
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 00:24:06 +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
stefan@webrtc.org
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
henrik.lundin@webrtc.org
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
stefan@webrtc.org
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
henrik.lundin@webrtc.org
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
andrew@webrtc.org
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
fischman@webrtc.org
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
solenberg@webrtc.org
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
sprang@webrtc.org
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
sprang@webrtc.org
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
pbos@webrtc.org
b613b5ab2b
Set local SSRC for VideoReceiveStream.
...
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.
BUG=2691
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 10:13:04 +00:00
henrik.lundin@webrtc.org
5ecdef11cc
Do not use recursive calling in NetEq test tools
...
This CL removes recursive calling in:
- NETEQTEST_DummyRTPpacket::readFromFile,
- NETEQTEST_RTPpacket::readFromFile.
The files currently exist for both NetEq3 and NetEq4, and all are
changed with this CL.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5200 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 08:26:49 +00:00
fischman@webrtc.org
e0034557a7
RTCPeerConnection(objc): avoid leaking ICE candidate on addition.
...
BUG=2670
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5199 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-02 18:49:54 +00:00
tina.legrand@webrtc.org
8418e9696b
Fixing NetEq tests for new Opus version
...
The new version of Opus doesn't generate the same number of bytes encoding the test vectors in audio_decoder_unittest. Therefore the test was updated not to check the length of the encoded packet, to prepare for the coming roll of Opus. Same change was applied to iSAC, which can also generate different number of bytes on different platforms.
BUG=1459
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5195 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-29 09:30:43 +00:00
braveyao@webrtc.org
54e8bfafba
Apprtc demo: add DSCP support.
...
BUG=2669
TEST=Manual Test
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5194 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-29 02:38:20 +00:00
phoglund@webrtc.org
03c7a35ac0
Fixing long lines in apprtc.py.
...
These long lines causes the presubmit to get angry.
BUG=webrtc:2678
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5193 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 17:45:08 +00:00
pbos@webrtc.org
e1fc3f22ea
Disable check for all sent SSRCs being valid.
...
Since the code for setting these up will set the codec before setting
SSRCs for the streams, any frames sent in between will be sent on
random-generated SSRCs.
This part should be added back during work on issue 1695.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 15:40:12 +00:00
bjornv@webrtc.org
bd41a84694
This CL adds an API to enable robust validation of delay estimates.
...
Added is
- a member variable for turning robust validation on and off.
- API to enable/disable feature.
- API to check if enabled.
- unit tests for these APIs.
Not added is
- the actual functionality (separate CL), hence turning feature on/off has no impact currently.
- calls in AEC and AEC, where the delay estimator is used. This is also done in a separate CL when we know if it should be turned on in both components.
TESTED=trybots, module_unittest
BUG=
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5191 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 14:58:35 +00:00
stefan@webrtc.org
b627f676b3
Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large.
...
BUG=2682
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5190 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 14:00:09 +00:00
pbos@webrtc.org
1f7c8d8b6a
Lock frame in ViECapturer::IncomingFrameI420.
...
r5160 explicitly assumed that IncomingFrameI420 was never called
sequentially. This assumption was found to be incorrect when some users
were changing beween existing capturers.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5189 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 13:26:33 +00:00
pbos@webrtc.org
13d38a13e3
Set up SSRCs correctly after switching codec.
...
Before SSRCs were not set up correctly, as the old VideoEngine API
doesn't support setting additional SSRCs before a codec with as many
streams are set.
No test was in place to catch this, so two tests are added to make sure
that we send the SSRCs that are set, and also that we can switch from
using one to using all SSRCs, even though initially not all of them are
set up.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5188 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 11:59:31 +00:00
bjornv@webrtc.org
d1a1c353ac
Recommit CL5184
...
TBR=aluebs@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/4599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5187 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 11:45:05 +00:00
solenberg@webrtc.org
c8f76ddc19
Refactor Remote Estimators Test into a more reusable form.
...
BUG=
R=andresp@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5186 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 10:11:35 +00:00
bjornv@webrtc.org
82eb3a690e
Revert 5184 "Small refactoring change in delay_estimator."
...
> Small refactoring change in delay_estimator.
>
> This CL produce the bit exact output and is a preparing step for an upcoming robust validation scheme.
>
> TESTED=trybots, module_unittest
> BUG=None
> R=aluebs@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4549004
TBR=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5185 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 09:44:47 +00:00
bjornv@webrtc.org
eea079a376
Small refactoring change in delay_estimator.
...
This CL produce the bit exact output and is a preparing step for an upcoming robust validation scheme.
TESTED=trybots, module_unittest
BUG=None
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5184 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 07:59:04 +00:00