89df092807
Make the destructor of AudioCodingModule public.
...
This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2200006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:27:43 +00:00
5eb997a2fd
Fix unsigned/signed comparison error due to r4729.
...
Fix it for real by switching to ints rather than casting.
TBR=xians
Review URL: https://webrtc-codereview.appspot.com/2191009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4730 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:01:42 +00:00
8f94013651
Reduce frequency of high audio delay warning logs.
...
This will log the warning every 5 seconds instead of every 10 ms.
BUG=b/10674993
TESTED=Ran voe_cmd_test with hard-coded high delay. Observed a log
every 5 seconds.
R=noahric@chromium.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4729 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 22:35:00 +00:00
256b83146c
Removes function that is not used anywhere but somehow still causing library load issues on Android Release build.
...
BUG=2364
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4728 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 20:43:13 +00:00
6138c5cfa4
OpenSl: fixes crashes externally reported in issue 2361 and 2362.
...
BUG=2361,2362
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4726 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 18:50:06 +00:00
036b7436df
Adding APIs. These APIs are not implemented yet, they are to help developement of ACM.
...
Un-implemented APIs.
TBR=henrik.lundin@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/2191008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4725 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 18:45:02 +00:00
d4d59ac871
Remove FrameForStorage:Follow up on r4688
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2201004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4723 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 15:18:15 +00:00
554d158ce6
Reset jitter buffer and timing if frames are getting too much delay.
...
BUG=chromium/263867
TEST=trybots
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2177005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4721 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 08:45:26 +00:00
835ef67d14
Remove repeated conditions key.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4720 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 00:16:00 +00:00
82f014aa0b
OpenSL (not default): Enables low latency audio on Android.
...
BUG=1669
R=andrew@webrtc.org , fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2032004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 18:24:07 +00:00
e07049f19f
Lock RTPSender statistics.
...
Suppressing these errors in TSan has become tedious. It's better to just
lock them.
BUG=2349
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2197004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 11:29:17 +00:00
eda189be14
Remove redundant STR_CASE_CMP macro definitions.
...
R=minyue@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2187005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 17:50:10 +00:00
021c42bfa8
Lock use of _packetRequestCallback in VCM.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4708 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 17:18:31 +00:00
59f20bb735
Break out RTCPSender dependency on ModuleRtpRtcpImpl.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 16:02:19 +00:00
86136a0e8f
Re-enable tests for Remote Bitrate Estimator
...
BUG=
R=stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4703 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 13:06:52 +00:00
0181b5f8dd
ExternalVideoDecoder for new VideoEngine API.
...
Implements the ExternalVideoDecoder interface for VideoReceiveStream.
Also adds a FakeDecoder used in tests, removing the overhead of running
the EngineTest tests with VP8 under Memcheck/TSan, allowing us to enable
them under Memcheck/TSan as well.
BUG=2346,2312
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2172004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4702 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 08:26:30 +00:00
30e055c4dd
Handle empty RTP video packets agnostic to codec.
...
Sending empty RTP packets caused a crash when using a generic codec
instead of VP8. This fix moves handling of empty RTP packets out of
ReceiveVp8Codec and into ParseRtpPacket.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2185004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4701 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-08 11:15:00 +00:00
b2c8a952a7
Improving padding rules and breaking out bw allocation to ViEEncoder.
...
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2170004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:58:01 +00:00
7bb8f02274
Adds support for combining RTX and FEC/RED.
...
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.
Enables retransmissions over RTX by default in the loopback test.
BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org , pbos@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2154004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:40:11 +00:00
5500d93fe5
Add temporal layer factory.
...
R=marpan@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2180004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4691 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 11:26:15 +00:00
f1e807c0e5
Removing FrameForStorage
...
R=pwestin@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2142004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 22:34:41 +00:00
bebf3995ce
Pre-multiply images for MouseCursorShape.
...
BUG=chromium:267270
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2173004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4685 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 19:32:46 +00:00
31b4a5ac82
Recognize armv7 target_arch for ios support in webrtc common.gyp
...
BUG=2343
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2176004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4684 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:46:36 +00:00
9080518a39
Restore severity precondition to logging.h.
...
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org , henrike@webrtc.org , kjellander@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:40:43 +00:00
164c4f71ba
Add clockdrift to RtpGenerator
...
RtpGenerator is a help class for NetEq testing. This change
add the possibility to simulate clockdrift. If no clockdrift is
set, the default is 0 (i.e., no drift).
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2175005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4680 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 12:16:38 +00:00
36439bf906
NetEq4: Small change to reduce allocs in AudioMultiVector
...
This change reduced the allocation count by 20000 in the bit-exactness
test.
BUG=Issue 1363
TEST=out/Debug/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2157004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4678 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 06:02:56 +00:00
77bf5c28c8
Clean capture timestamp code.
...
BUG=
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2134004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4675 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:35:43 +00:00
b21e528c60
Protecting Bitrate to avoid data race found by tsan.
...
TEST=try and vie_auto_test with tsan.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2163004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4673 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 08:42:44 +00:00
65abb6b1ed
Revert 4671 "Enable SetInitialPlayoutDelay on Android."
...
Tests enabled in r4671 failed:
build.chromium.org/p/client.webrtc/builders/Android%20Tests/builds/31/steps/slave_steps/logs/stdio
> Enable SetInitialPlayoutDelay on Android.
>
> Background:
> In Chrome mirroring which uses 500ms buffering mode,
> audio video mismatch happens in the begining because of the lack of the api.
>
> BUG=b/10538425
> TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2144004
TBR=dwkang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4672 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 07:47:39 +00:00
310ac91d2a
Enable SetInitialPlayoutDelay on Android.
...
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.
BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2144004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4671 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 01:19:12 +00:00
3c5a9242fe
Don't force cont' when enabling kWithErrors
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2047004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4669 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 20:45:36 +00:00
2b810bf77b
Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2143004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4667 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 19:09:49 +00:00
cac7325b84
Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule.
...
Found with tsan.
TEST=try job and tsan
R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/2156004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4661 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 12:11:12 +00:00
8fb89533af
Correcting two nits in InputAudioFile
...
First, the fread function returns number of elements read, not
necessarily the number of bytes. In this case, it is the number
of samples. Second, a spelling mistake was corrected.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2161004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4658 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 08:43:28 +00:00
b3e905cd91
Disable all LS_VERBOSE logging in NetEq4
...
This reduces exectution time of NetEqDecodingTest.TestBitExactness
with almost 30% and reduces the allocation count (from valgrind)
with almost 50% for the same test.
An issue has been created to re-enable logs when logging performance
is improved; see https://code.google.com/p/webrtc/issues/detail?id=2317 .
BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2136004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4652 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 09:41:06 +00:00
c487c6abb0
NetEq4: Make the algorithm buffer a member variable
...
This reduces the alloc count by more than 100,000 for
NetEqDecodingTest.TestBitExactness.
BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4651 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 07:59:30 +00:00
45d2840623
Zero comfort noise for stereo insted of assertion.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2084004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4645 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 15:37:08 +00:00
3170b5750f
Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics().
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2107005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4644 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 15:36:53 +00:00
9ded07e3a4
Fix typo in InvertedDesktopFrame
...
BUG=279334
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2141004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4643 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 01:05:14 +00:00
d7301775f5
update neteq 4 to facilitate NACK
...
BUG=
R=turaj@webrtc.org , turajs@google.com
Review URL: https://webrtc-codereview.appspot.com/2008004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4637 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 00:58:14 +00:00
e141373b8a
Add isolate configuration for Android for all tests.
...
In https://code.google.com/p/webrtc/source/detail?r=4407
henrike@ added the path to the WebRTC resources and
data directories for Android that are required in order to
use isolate for test execution on Android.
This CL adds similar entries to the rest of the .isolate
files added in
https://code.google.com/p/webrtc/source/detail?r=4590 .
It also removes three accidentally added .isolate files that originated
from old test names:
* audio_device_test_api
* video_capture_module_test
* video_render_module_test
BUG=1882,1916
TEST=trybots passing.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2107004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4627 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 12:10:09 +00:00
ee92b664b3
Re-organizing ACM tests
...
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret.
While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL.
I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD().
BUG=issue2173
R=minyue@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1961004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 07:33:51 +00:00
01cb3ad883
Fix image flipping for OpenGL-based screen capturer on Mac.
...
I broke captured image flipping when refactoring this code while it was
still in chromium. Previously we had CaptureData that was returned from
capturers with correctly inverted stride, but frames were still stored
with positive stride. CaptureData was removed and so the returned frames
always had positive stride, which is not correct. Now ScreenCapturerMac
uses frames with inverted stride when capturing using OpenGL.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2105004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4621 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 21:48:56 +00:00
e3de6b1e90
Enable ObjC build by default and reenable 64-bit mac libjingle build
...
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2080004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 19:31:21 +00:00
f31a47abdc
VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that,
...
BUG=
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/2077004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4614 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:10:11 +00:00
b2c28c3699
Relanding 4597 - Don't force key frame when decoding with errors.
...
Makes sure that incomplete key frame or delta frames will be released from the JB when decoding with errors.
The decoder in turn will trigger a PLI until a complete key frame is received in order to start a session.
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/2097004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4607 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 21:54:50 +00:00
9f282403f2
WindowCapturer implementation for Linux.
...
Window enumeration is based on the code used by hangouts plugin
(see libjingle/talk/base/linuxwindowpicker.cc). XServerPixelBuffer
is used to capture windows. It had to be refactored to support window
capturing (previously it worked only for the whole screen).
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1741004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4605 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 18:22:12 +00:00
ceea41d135
Revert 4597 "Don't force key frame when decoding with errors"
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> Don't force key frame when decoding with errors
>
> BUG=2241
> R=stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2036004
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2093004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4600 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 00:53:24 +00:00
eef29ec6cf
Implement window capturer for OS X.
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R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2055005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4599 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 00:39:46 +00:00
44af55cc44
Don't force key frame when decoding with errors
...
BUG=2241
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2036004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4597 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 23:29:43 +00:00
