c279a5d72c
Wire up RTX in VideoReceiveStream.
...
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.
BUG=2399
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 09:30:53 +00:00
8d375c95b7
Fix deadlock on register/unregister observer while there is a an going callback.
...
BUG=2835
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 23:09:25 +00:00
754de528b7
Fix array declarations in aec_rdft.h.
...
Was causing warnings in Chromium such as:
warning C4742: 'rdft_wk2i' has different alignment in
'webrtc\modules\audio_processing\aec\aec_rdft_sse2.c' and
'webrtc\modules\audio_processing\aec\aec_rdft.c': 4 and 16
BUG=chromium:336620
R=cduvivier@google.com
Review URL: https://webrtc-codereview.appspot.com/7489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5419 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 20:55:14 +00:00
e7223e7795
Set NACKed packet to -1 in TestNackRetransmission.
...
Zero is a valid sequence number which may occur even if there are no
retransmissions, this caused the test to flake as an incoming packet
would be mistaken for a retransmission.
BUG=2830
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5417 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 16:14:34 +00:00
0e93257cee
Add callbacks for receive channel RTP statistics
...
This allows a listener to receive new statistics (byte/packet counts, etc) 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=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 10:00:39 +00:00
91db93d24f
Android, fixes crash on devices with only front cameras.
...
BUG=2807
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5415 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-22 21:31:24 +00:00
7de3bb9df9
Output logs to stderr from voe_cmd_test by default.
...
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.
TESTED=logs output to stderr by default and to the usual file when the
flag is specified.
R=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 22:17:43 +00:00
28da47c52f
Android example apps: fixes issue where useful failure information was suppressed.
...
BUG=2808
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 19:03:51 +00:00
7dba27c740
Potential dead lock in receive statistics
...
A dead lock could occur if the following to code paths are called
concurrently:
ReceiveStatisticsImpl::IncomingPacket() ->
StreamStatisticianImpl::IncomingPacket()
StreamStatisticianImpl::GetStatistics() ->
ReceiveStatisticsImpl::StatisticsUpdated()
Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().
BUG=2818
R=asapersson@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 16:33:37 +00:00
32c3247418
Fix for libtalkmobile build error
...
bug=b/12549061
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 16:16:58 +00:00
7ef7df57d8
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5403 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 15:54:56 +00:00
e02d47515f
Set up receiver RTX config using a std::map.
...
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 14:43:55 +00:00
efaeda0c76
Add configuration and test for extended RTCP reference time reports to new video api.
...
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 08:34:49 +00:00
32c26eb90b
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
...
BUG=N/A
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 23:12:51 +00:00
4985927d36
Implement screen enumeration and individual screen capturing for Windows.
...
BUG=2787
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5399 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 17:19:16 +00:00
ead202b973
Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started.
...
BUG=2801
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 23:26:37 +00:00
0af1ffa84d
Android, WebRTCDemo: fix issue where changing remote IP was not working properly.
...
BUG=2783
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 16:45:15 +00:00
4ffd9c7423
Add full path to headers
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 09:01:39 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
