b6541ca3a1
Ensure capture_levels_ is sized correctly at init time.
...
Fixes failing voe_auto_test and audioproc_perf.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/6699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5348 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 18:36:10 +00:00
cf9d364063
Now printing less output from compare_videos.py.
...
Alternative solution to the one in
https://codereview.chromium.org/114003006/ .
I considered adding a verbose flag, but it needs to be passed through
like 5 functions, so I didn't think it was worth it for a function of
such speculative use.
BUG=chromium:327990
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5347 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 17:59:30 +00:00
60730cfe3c
Remove the requirement to call set_sample_rate_hz and friends.
...
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org , bjornv@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 17:45:09 +00:00
39669c5c8f
Remove outdated DestroyVideoSendStream comment.
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5345 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 12:27:22 +00:00
ccd42840bc
Wire up statistics in video send stream of new video engine api
...
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5559006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 09:54:34 +00:00
0b7d8e6fcb
AppRTC: Alert the user to failure to acquire TURN server.
...
Hopefully will result in quicker turnaround time for CEOD/turnserver fixes.
Might trigger undesirable levels of bogus/spammy/unhelpful/PEBCAK reports to
discuss-webrtc, in which case I'll remove the second part of the message.
R=juberti@google.com , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4779005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5343 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-06 23:46:53 +00:00
acc05ac7d1
Roll libvpx 232686:241571
...
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/6599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5342 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-06 21:04:22 +00:00
a9bdee6690
Add Christophe Dumez to AUTHORS.
...
Copied from Chromium's AUTHORS.
R=ch.dumez@samsung.com
Review URL: https://webrtc-codereview.appspot.com/5559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5341 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-06 19:43:21 +00:00
7bdaf837d4
Updated PeerConnection samples so they run on FF.
...
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5340 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-03 23:13:01 +00:00
f6d6ed0c66
Update talk to 59039880.
...
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5339 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-03 22:08:47 +00:00
e667234ee2
libyuv r949 includes changes to allow any width, mainly relating to fixed point math overflows.
...
BUG=none
TEST=try bots
R=ronghuawu@google.com
Review URL: https://webrtc-codereview.appspot.com/6579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5338 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-03 18:57:22 +00:00
a89d17d5b7
Delay Estimator: robust_validation should be stored over a reset
...
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-02 07:07:04 +00:00
2240763ec2
libyuv r930 for RGB24ToUV_NEON improved color accuracy to avoid red tint, and use malloc with variable sized row buffers to avoid stack overflow and relax width restrictions. Previously was limited to 4k on x86 and 1080p on arm. In practice the new limitation is 32767 pixels wide.
...
BUG=none
TESTED=try bots
R=tpsiaki@google.com , wjia@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-28 07:00:18 +00:00
2fb72cfeec
Add include guards to forward_error_correction_internal.h
...
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5789005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5335 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-24 05:06:12 +00:00
0062a6d099
Fix the include guard in transmit_mixer.h
...
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5334 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-24 03:58:51 +00:00
a7cfa6704a
Fix the include guard in transmit_mixer.h
...
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5333 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-24 03:39:10 +00:00
000dde99c8
Android build: make it quiet on success and not overly noisy on failure.
...
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 22:49:35 +00:00
a63fc87139
Fix JS error in adapter.js for FF for the case when ?transport=xxx is missing in TURN url.
...
BUG=2737
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5331 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 22:10:17 +00:00
f6acf98a46
Fix the android clang bot for compiling with thread annotations.
...
TBR=niklas.enbom@webrtc.org
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6279005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5330 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 21:54:26 +00:00
cf2b3acc48
Update Android trybots in the default try job list.
...
This updates the default set of trybots that are used
when no bot names are specified when submitting a try job.
TBR=andrew@webrtc.org
TEST=Ran git try -t compile and verified it was sent to all bots.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/6289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5329 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 21:20:42 +00:00
7fb75ecbd4
Add thread_annotations for clang targets.
...
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.
R=niklas.enbom@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 20:20:50 +00:00
6031001565
If the configured start bitrate is higher than the configures max
...
bitrate, cap the star rate accordingly.
BUG=2720
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5327 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 15:07:12 +00:00
8dbca8d665
Race condition in ViECapturer::RegisterObserver
...
Critical section ViECapturer.observer_cs_ should be taken when
registering an observer.
BUG=2734
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5326 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 11:36:03 +00:00
a463d73b99
Update WebRTC to version 3.48
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5324 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 22:38:38 +00:00
54ae4ffb9e
Add callbacks for receive channel RTCP statistics.
...
This allows a listener to receive new statistics 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=henrika@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 13:26:02 +00:00
e682aa5077
Refactoring MediaOptimization so it can easily be turned into a thread-safe class.
...
BUG=2732
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5322 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 10:59:48 +00:00
faada6e604
Integrate fake_network_pipe into direct_transport.
...
TEST=trybots
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 20:28:25 +00:00
8f99a18119
Port scale and compare functions to pepper_33 and mips.
...
BUG=none
TEST=validator passes with new toolchain.
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5320 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 19:51:37 +00:00
5fe2d65c43
Remove metrics_unittests
...
This target has been merged into video_engine_tests in r5284.
BUG=webrtc:1843
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5319 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 13:27:37 +00:00
8a54417968
Remove media_file from VideoEngine dependencies.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 10:00:29 +00:00
b429e516a9
cpplint cleaning new API and its implementation files.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:46:22 +00:00
bcd124cdba
Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand().
...
Follow up steps is to support NackConfig.rtp_hostory_ms and/or increase fake encoder bitrate.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5316 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:45:45 +00:00
1fa41be66a
Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5315 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:44:53 +00:00
8ae72560dd
Make MouseCursor mutable
...
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.
R=dcaiafa@chromium.org
Committed: https://code.google.com/p/webrtc/source/detail?r=5310
Review URL: https://webrtc-codereview.appspot.com/6059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5314 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 02:18:01 +00:00
f8be8df33a
audio_processing_unittest: unbreak clang compilation.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5313 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:39 +00:00
179908c81c
JNI Audio: remove dead members.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:14 +00:00
e4c927208b
Revert "Make MouseCursor mutable"
...
This reverts commit a6db8ab8bc4b569a26633b0ca3665297f1a5349b.
TBR=dcaiafa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5311 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 22:48:50 +00:00
8fd1d26536
Make MouseCursor mutable
...
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.
R=dcaiafa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5310 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 22:19:12 +00:00
86ba294d25
am 148d5389: Merge "AArch64: Add AArch64 to webrtc typedefs"
...
* commit '148d5389f1c24942f0c8e5d011a2c8d123f817c9':
AArch64: Add AArch64 to webrtc typedefs
2013-12-17 22:17:40 +00:00
148d5389f1
Merge "AArch64: Add AArch64 to webrtc typedefs"
2013-12-17 22:03:06 +00:00
af320fd2f7
The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6019004
Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 21:33:27 +00:00
50f7b2da5d
roll libyuv to r915 for webview jpeg build fix and NaCL pepper_33 initial support.
...
BUG=none
TEST=try bots
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 18:18:17 +00:00
504679fd92
AArch64: Add AArch64 to webrtc typedefs
...
Change-Id: I34ca53b1dd3d56b8df6f7c5440e2cf49db4c7546
Signed-off-by: Craig Barber <craig.barber@arm.com >
2013-12-17 13:33:06 +00:00
052fa6243a
Stop transport in test SuspendBelowMinBitrate.
...
Avoids race when packets are still left in the network while the Call is
being destroyed.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/6009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5307 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 11:19:58 +00:00
e6b871bb29
Added method for getting default module state and protect agains a
...
read/write race for child_modules_.
BUG=2731
TEST=tsan
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 08:30:40 +00:00
9df6674b26
Scale down by 4x with box filter. Fix for 1 pixel wide bilinear filter. Fix for I420ToARGB overread on V plane that causes valgrind fail.
...
BUG=none
TESTED=gcl try libyuv_r911 --bot=linux_valgrind
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5305 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 08:25:31 +00:00
eb7b7bce3d
Modify video_render/ to allow a single old frame.
...
This stabilizes tests as a single frame reaches end-to-end, as well as
allowing slow or heavily-loaded systems to see any video updates even if
the frame takes more than 500ms in the pipeline.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=2724
Review URL: https://webrtc-codereview.appspot.com/5949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5303 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 18:24:37 +00:00
5b3c67ef25
objc/README: Remove outdated advice about target_os.
...
BUG=chromium:248168
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5979005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5302 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 17:15:19 +00:00
919f87fb36
Delete capturers after destroying streams in test.
...
Since the renderers in CallTest.SendsAndReceiveStreams also stopped the
capturers they must be deleted after the VideoReceiveStream is stopped
or an use-after-free may occur.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5300 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:55:54 +00:00
e7b1e11283
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
...
> 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
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:40:36 +00:00
