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Author SHA1 Message Date
bjornv@webrtc.org
6a94734d4d Adds back set_sample_rate_hz() when Init is called in recordings. 
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.

BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 08:41:09 +00:00
andrew@webrtc.org
ea9392d5eb MIPS optimizations for NS audio processing module 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139006

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 07:22:01 +00:00
sergeyu@chromium.org
fb4e256d49 Fix crash in MouseCursor::CopyOf() 
This issue was causing test failures with the latest webrtc roll.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 04:45:35 +00:00
andrew@webrtc.org
8f35afab8c Exclude protoc objects from merge_libs.py. 
BUG=b/12567343
R=wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 00:31:57 +00:00
mallinath@webrtc.org
7a2ca7c621 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. 
This change is also must for rolling webrtc in chrome.

R=jiayl@webrtc.org
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 19:00:13 +00:00
jiayl@webrtc.org
017b619010 Extends the ScreenCapturer interface for individual display screen cast. 
Real implementations for each platform will be added in future CLs.

BUG=2787
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/6819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 18:26:37 +00:00
wjia@webrtc.org
03cfde2d10 Roll Chromium 238260 -> 243863 
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:48:34 +00:00
pbos@webrtc.org
39fcfd78ae Remove empty VideoCodecGeneric struct. 
Struct was added prematurely and triggers a warning with
-Wextern-c-compat in latest clang.

R=henrika@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/7119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 12:55:59 +00:00
henrik.lundin@webrtc.org
d9faa46d57 Changing to using factory methods for some classes in NetEq 
In this CL, the Expand, Accelerate and PreemptiveExpand objects are
created using factory methods. The factory methods are injected into
NetEqImpl on creation. This is a step towards implementing a no-decode
operation.

BUG=2776
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 10:18:45 +00:00
aluebs@webrtc.org
4371d4650a Temporarily disabling some more audio processing tests. 
R=andrew@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 08:57:22 +00:00
sergeyu@chromium.org
eb31b45aaf Fix MouseCursorMonitorMac to return correct hotspot position. 
Previusly (0, 0) was always return as mouse cursor hotspot.

BUG=2779
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 23:25:17 +00:00
henrike@webrtc.org
3907c2e7e5 Removes the remaining uses of the list wrapper class and the list wrapper class. 
BUG=2164
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:41:34 +00:00
fischman@webrtc.org
dde7aee40f WebRTCDemo: fix out-of-bounds array read. 
Also removed the WebRtcCamera class, which has become an empty wrapper around
CameraInfo in the post-rewrite world.

First pointed out by Jeremy Mao <yujie.mao@webrtc.org> in
http://review.webrtc.org/6869004/

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:15:38 +00:00
elham@webrtc.org
ad1863de74 Updated Webrtc version to 3.49 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 17:49:49 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files. 
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 15:21:30 +00:00
andresp@webrtc.org
d0b436a935 Revert "Activate ACM test for Android in modules_tests." (rev5364). 
TBR=turaj@webrtc.org,tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 13:15:59 +00:00
aluebs@webrtc.org
8bc4fcfeb6 Temporarily disabling audio processing tests. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6889005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:14:47 +00:00
henrik.lundin@webrtc.org
2c03bf1641 Increasing simulation time for NetEqPerformanceTest 
This is to get better "signal-to-noise ratio" in the performance bots.
The neteq4-runtime metric is expected to increase by a factor of 10.

BUG=2397
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:04:23 +00:00
bjornv@webrtc.org
bbd47fc5b5 Enables robust delay validation in AEC delay logging. 
* Explicitly disabled robust validation in AECM.
* Updated audio_processing_unittests for using robust delay validation in AEC.
* Updated output_data_float.pb (not needed for Android nor fixed point, since AECM is untouched).

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 08:54:34 +00:00
andrew@webrtc.org
023cc5abc7 Minor voice engine improvements around AGC. 
- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.

R=aluebs@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-11 01:25:53 +00:00
henrike@webrtc.org
573a1b45b5 Android: Fixes crash when exiting WebRTCDemo. 
BUG=2738
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:58:06 +00:00
turaj@webrtc.org
7cc64b3747 Activate ACM test for Android in modules_tests. 
TEST=local on Nexus 7.
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:35:09 +00:00
pbos@webrtc.org
f777cf2547 Permitting double start/stopping of streams. 
It doesn't make too much sense to hard enforce that the user keeps track
of which streams are started and which are not.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 18:47:32 +00:00
henrik.lundin@webrtc.org
a366e810a9 Adding NetEq performance test to webrtc_perf_tests 
The performance test is based on the neteq4_speed_test application. The
bulk of the test code is extracted into a test class, and included into
the neteq_unittest_tools target. The actual gtest that runs the
performance test is implemented in neteq_performance_unittest.cc, and
built as a part of webrtc_perf_tests.

The old stand-alone test application is now made dependent on the new
test class, to avoid code duplication.

BUG=2397
R=andrew@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 08:24:04 +00:00
bjornv@webrtc.org
fa8d534e09 Delay Estimator: Adds unittests for robust validation. 
In addition to unittests a cast losing constness was corrected.
The tests added are:
1. Adjusting allowed_offset when robust validation is disabled should have no impact.
2. For noise free signals there should be no difference between robust validation or not.
3. Robust validation acts faster during startup.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 07:42:07 +00:00
henrik.lundin@webrtc.org
e7ce437333 Fixing lint errors in NetEq4 
Just taking care of a few old lint errors.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 14:01:55 +00:00
andresp@webrtc.org
c5aeb2aa15 Make code simpler on VCMEncodedCallback. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:04:32 +00:00
andresp@webrtc.org
1df9dc3957 Isolate register post encode callback in video coding module to simplify code and critical sections. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:01:57 +00:00
andresp@webrtc.org
b08a12d6e8 Isolate debug recording from video sender into a thread safe small class. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 12:38:22 +00:00
solenberg@webrtc.org
ab2405164a Add another test case for AST/TOF switching. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:59:44 +00:00
bjornv@webrtc.org
bccd53de57 Delay Estimator: Converts a constant into a configurable parameter. 
The parameter is used in the robust validation scheme, which will be turned on in a separate CL.

* Setter and getter for allowed delay offset.
* Updated unittests.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:18:15 +00:00
andrew@webrtc.org
d335094852 Init to 16 kHz in the fixed-point profile. 
Fixes modules_unittests for fixed-point builds (Android).

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:57:10 +00:00
andrew@webrtc.org
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
phoglund@webrtc.org
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
andrew@webrtc.org
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
pbos@webrtc.org
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
sprang@webrtc.org
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
bjornv@webrtc.org
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
braveyao@webrtc.org
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
braveyao@webrtc.org
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
braveyao@webrtc.org
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
fischman@webrtc.org
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
andresp@webrtc.org
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
andresp@webrtc.org
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
mflodman@webrtc.org
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
sprang@webrtc.org
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
tnakamura@webrtc.org
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
sprang@webrtc.org
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
andresp@webrtc.org
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
stefan@webrtc.org
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