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Author SHA1 Message Date
pbos@webrtc.org
432893a100 Remove TODO saying to remove WebRtcVideoFrame. 
Comment was added prematurely, there's no decision to get rid of this
type.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 21:17:22 +00:00
pbos@webrtc.org
b15dddf7ae Remove files from talk/PRESUBMIT.py blacklist. 
Many files can now be submitted here and do not have to be rolled in.

BUG=
R=henrike@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6893 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 20:38:53 +00:00
henrike@webrtc.org
d968dd039a Fixes failure triggered by include order re-ordering. 
BUG=N/A
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 18:39:43 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 17:26:08 +00:00
marpan@webrtc.org
b242c44354 Further DrMemory suppressions, likely from r6811 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 17:17:40 +00:00
buildbot@webrtc.org
2c0fb05f16 (Auto)update libjingle 73221069-> 73222930 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 16:47:12 +00:00
buildbot@webrtc.org
67f849575c (Auto)update libjingle 73215194-> 73221069 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 16:22:04 +00:00
henrike@webrtc.org
4eeeefebb2 (Auto)update libjingle 73072800 -> 73215194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 14:57:30 +00:00
asapersson@webrtc.org
23a4d8522e Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4). 
Increased kStandardRampUpDelayMs (30 to 40s).

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 14:33:49 +00:00
xians@webrtc.org
38d88816e3 Fix the audio source failure due to unsupported constraints. 
Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.

This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.

BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback

R=henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 13:51:58 +00:00
henrik.lundin@webrtc.org
5af76aedcd Removing TODOs related to AcmReceiverBitExactness checksums 
Should have been part of r6883.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 13:02:00 +00:00
henrik.lundin@webrtc.org
388bd79a76 Update checksums for AcmReceiverBitExactness on android 
This should have been a part of r6882.

BUG=3519
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 10:38:15 +00:00
henrik.lundin@webrtc.org
023f12fb6e NetEq background noise generation off by default 
This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.

With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.

Reference files are updated where needed.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 09:45:40 +00:00
stefan@webrtc.org
c27543d297 Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field. 
BUG=3679
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 07:40:45 +00:00
mallinath@webrtc.org
e999bd087b Removing ASSERT for tcp candidate for port 0 and 9, as Android clients 
may not be called with set_allow_tcp_listen(false).

This CL will also sends tcp candidate in RFC 6544 format.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 06:05:55 +00:00
pbos@webrtc.org
afb554f404 Move default-recv-channels to a separate class. 
BUG=1788,3099
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 23:17:13 +00:00
fbarchard@google.com
c891fee7ab Make a int64 constant use ULL suffix so it wont get truncated. 
BUG=3690
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 22:39:06 +00:00
marpan@webrtc.org
c6273b53dd DrMemory suppresssions, likely from r6811. 
BUG=3655
R=henrike@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 21:29:06 +00:00
pbos@webrtc.org
c3d2bd28a3 Fix GetStats() crash. 
GetStats() can be called before codecs are set and the underlying
webrtc::VideoSendStream is created, leading to a null-pointer
dereference.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 20:55:10 +00:00
henrike@webrtc.org
3d53f614bd .gitignore removed openssl 
BUG=N/A
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/19029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 16:04:00 +00:00
henrike@webrtc.org
aa2344e741 talk/third_party: removes the empty directory. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 15:57:02 +00:00
buildbot@webrtc.org
8d57f08902 (Auto)update libjingle 73072800-> 73072800 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 14:41:46 +00:00
phoglund@webrtc.org
40995c7fd0 Fixing uninitialized variable in file_audio_device.cc. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 11:09:12 +00:00
bjornv@webrtc.org
0a3cbb3906 common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32 
The macro is only used at four places in iSAC fixed point and the macro have been replaced at those places.
In addition, it is used in a unit test, but throws a warning treated as error (issue3674).

The macro has both MIPS and armv7 optimizations. Removing them impacts only MIPS platforms without DSP ASE. This may cause a very small increase in complexity when using iSAC fix.
The armv7 optimizations are not used anywhere, since specific ones are used inline in iSAC fix.

BUG=3348,3353,3674
TESTED=locally and trybots
R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 10:54:50 +00:00
bjornv@webrtc.org
cf8f33a6d6 Removes mismatching signs in signal_processing_unittests 
Negative inputs was used in WebRtcSpl_NormU32() causing warnings.

BUG=3674
TESTED=locally and trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 10:27:21 +00:00
minyue@webrtc.org
6aac93bd9c Adding SetOpusMaxBandwidth in VoE and ACM 
This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.

TEST = added a test in voe_cmd_test and listened to the result

BUG=
R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 08:13:33 +00:00
bjornv@webrtc.org
c98ce3b34c modules/audio_processing: Updates output_data_fixed.pb test file 
In r6591 a shift macro was removed affecting AECM. In addition to that change a bug was fixed. The fix added a few voice_counts in ApmTest.Process.

This CL updates the reference file, even though it is not used in practice since the test is currently turned off for Android (where AECM is used).

BUG=3672
TESTED=locally
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 07:35:52 +00:00
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
bjornv@webrtc.org
820f8e9ca7 modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples 
audio_processing did not compile when aec_untrusted_delay_for_testing=1 was set. The constant kDelayDiffOffsetSamples was declared only for Mac when WEBRTC_UNTRUSTED_DELAY was automatically turned on.

Moving the declaration outside the ifdef makes it build with the flag on for any platform.

BUG=3673
TESTED=locally and trybots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 15:39:00 +00:00
henrik.lundin@webrtc.org
4e4b0984da Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics 
The two tests both read and process the same (rather long) RTP input
file, and simply look at different outputs. This change merges the two
tests into one, in order to reduce testing time.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:48:49 +00:00
henrike@webrtc.org
065247b5b7 Rebase webrtc/base with r6863 version of talk/base: 
cls integrated: r6809
svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base > 6809.diff
patch -p0 -i 6809.diff

BUG=3379
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:32:13 +00:00
tommi@webrtc.org
730bf30da7 Refactor StatsCollector and associated types. 
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:08:33 +00:00
henrik.lundin@webrtc.org
1c8391205e Use test::Packet test::PacketSource classes in neteq_rtpplay 
This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:29:38 +00:00
bjornv@webrtc.org
96d8b0e69f Revert 6860 "SSE2 version of SubbandCoherence()" 
> SSE2 version of SubbandCoherence()
> 
> The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
> reported by audioproc is ~3.3%
> 
> The output is bit exact.
> 
> R=bjornv@webrtc.org, cd@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18779004
> 
> Patch from Scott LaVarnway <slavarnw@gmail.com>.

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:09:13 +00:00
bjornv@webrtc.org
0db82f337f SSE2 version of SubbandCoherence() 
The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
reported by audioproc is ~3.3%

The output is bit exact.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18779004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 10:38:31 +00:00
jiayl@webrtc.org
7ec3f9f838 Fix a bug in parsing IceCandidate with IPV6 address. 
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.

BUG=3669
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 23:09:15 +00:00
buildbot@webrtc.org
9eabe5e912 (Auto)update libjingle 72931377-> 72931377 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 22:48:28 +00:00
mallinath@webrtc.org
2d60c5e8bc Encoding and Decoding of TCP candidates as defined in RFC 6544. 
R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204

Review URL: https://webrtc-codereview.appspot.com/21479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 22:29:20 +00:00
harryjin@google.com
8c01e59424 Allow root build dependencies to be overridden. 
R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/22039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 00:08:58 +00:00
buildbot@webrtc.org
53df88c1bc (Auto)update libjingle 72847605-> 72850595 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:46:01 +00:00
buildbot@webrtc.org
65b98d12c3 (Auto)update libjingle 72839629-> 72847605 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:09:08 +00:00
henrike@webrtc.org
3763b9bda0 webrtc/base: removes linkage of crypto 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 21:26:18 +00:00
tkchin@webrtc.org
c8554be6dd Support for TURN/TLS. 
Wrap the socket in an SSL adapter, then simply call StartSSL() on the
SSLAdapter instance.

Cloned from: https://webrtc-codereview.appspot.com/21799004/

R=juberti@chromium.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/14059004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:39:08 +00:00
tkchin@webrtc.org
cb46de24fb Add new OWNERS file to talk/examples. 
R=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:01:34 +00:00
buildbot@webrtc.org
5b1ebacca2 (Auto)update libjingle 72820109-> 72822008 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 17:18:00 +00:00
buildbot@webrtc.org
d509678a4e (Auto)update libjingle 72819313-> 72820109 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:57:07 +00:00
buildbot@webrtc.org
94b996cc18 (Auto)update libjingle 72785516-> 72819313 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:47:14 +00:00
stefan@webrtc.org
59a2f9f584 Remove the old H264 code now that a new H.264 packetizer has been implemented. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:09:24 +00:00
stefan@webrtc.org
9d74f7ce8c Fix single nalu packetization bug. 
Nalus which had the same size as the max payload size would cause the payload size accounting to wrap.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:02:16 +00:00
pbos@webrtc.org
e8c84bf4de Fix so video_replay logs aren't spammed. 
Add unknown-SSRC counters instead and log number of unknown packets at
end of session.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/13119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 14:42:45 +00:00