fc10c5c97f
This is related to https://code.google.com/p/webrtc/issues/detail?id=1341
...
BUG=1341
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2225004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4761 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 05:15:37 +00:00
e6ac163145
This is related to https://code.google.com/p/webrtc/issues/detail?id=846
...
BUG=846
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2224004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4760 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 05:15:19 +00:00
e401c2e391
Split video coding module unit tests into sender and receiver unit tests.
...
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-16 20:29:13 +00:00
8fa436bd65
Remove use of vcm->ResetDecoder from modules/utility.
...
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2203006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4750 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-16 11:26:35 +00:00
f7eb75be1a
Split VideoCodingModuleImpl into VideoSender and VideoReceiver.
...
Only implmentation is changed the interface to the module is unchanged for now.
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2200008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4746 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-14 00:25:28 +00:00
48af652ea5
Prepare to compile ACM1 and ACM2.
...
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2206004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 23:06:59 +00:00
6ab45b9dab
Implement DesktopRegion subtraction.
...
Region subtraction is used in chromoting client, so it's needed to
replace SkRegion.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2205004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4741 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 19:53:16 +00:00
1f09dbe353
Moving test-only code (stream_generator) out of vcm implemention.
...
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2207004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4740 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 19:17:54 +00:00
2553450959
Fix win trybot errors due to r4729.
...
The addition of logging.h in r4729 was causing the win trybot to fail
with "#pragma deprecated" errors in standard library headers. This
turned out to be due to including strsafe.h (via audio_device_config.h)
before sstream (via logging.h).
strsafe.h was only being included for the unused DEBUG_PRINT macro. I
removed all references to it.
This incidentally removes a bunch of other unneeded headers discovered
while trying to track the problem down.
This didn't show up in the commitbots; my guess is that the trybots are
using the VC10 toolchain and the commitbots the VC11 toolchain.
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/2204004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4738 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 00:02:13 +00:00
6a5cc9d899
Fix crash in the window capturer on windows
...
BUG=crbug.com/289753
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2203005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4737 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 19:17:26 +00:00
7959e16cc2
ACM2 integration with NetEq 4.
...
nack{.cc, .h, _unittest.cc} are basically copies from main/source/ folder, with cpplint warning cleaned up.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4736 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 18:30:26 +00:00
82a846f0cb
Adding Ami to the video renderer and capturer modules.
...
TBR=fischman@webrtc.org ,wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2202006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4735 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:43:17 +00:00
36cf4d2309
The video render module for iOS.
...
BUG=2105, 2028
R=fischman@webrtc.org , mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2064004
Patch from SeungJae Lee <sjlee@webrtc.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4734 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:39:53 +00:00
e509f943ed
This issue is related to
...
https://chromereviews.googleplex.com/9908014/
I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.
BUG=
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2171004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:03:00 +00:00
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
