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Author SHA1 Message Date
andresp@webrtc.org
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
turaj@webrtc.org
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
sergeyu@chromium.org
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
andresp@webrtc.org
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
andrew@webrtc.org
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
sergeyu@chromium.org
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
turaj@webrtc.org
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
mallinath@webrtc.org
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
fischman@webrtc.org
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
minyue@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
henrike@webrtc.org
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
henrike@webrtc.org
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
turaj@webrtc.org
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
mikhal@webrtc.org
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
stefan@webrtc.org
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
andrew@webrtc.org
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
henrike@webrtc.org
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
pbos@webrtc.org
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
andrew@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
solenberg@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
andresp@webrtc.org
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
mikhal@webrtc.org
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
alexeypa@chromium.org
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
fischman@webrtc.org
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
andrew@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
andresp@webrtc.org
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
mflodman@webrtc.org
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
mflodman@webrtc.org
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
dwkang@webrtc.org
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
mikhal@webrtc.org
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
mikhal@webrtc.org
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
mflodman@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
turaj@webrtc.org
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
turaj@webrtc.org
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
sergeyu@chromium.org
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
minyue@webrtc.org
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