4636 Commits

Author SHA1 Message Date
Henrik Lundin
82d6f2a3f7 ACM: Remove ACMVQMonCallback object
It was never used, and the underlying functionality was removed long
ago.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1365193003 .

Cr-Commit-Position: refs/heads/master@{#10083}
2015-09-28 08:25:33 +00:00
henrika
69984f0533 Fixes logging levels in WebRtcAudioXXX.java classes
BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1363673005 .

Cr-Commit-Position: refs/heads/master@{#10082}
2015-09-28 07:24:16 +00:00
Henrik Kjellander
d6d27e7340 Update isolate.gypi to support Swarming + move .isolate files
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)

Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.

In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.

BUG=497757
R=maruel@chromium.org
TBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc

Review URL: https://codereview.webrtc.org/1373513002 .

Cr-Commit-Position: refs/heads/master@{#10081}
2015-09-25 20:19:21 +00:00
deadbeef
c97be6a741 Disable TestUdpReadyToSendIPv4 under MSan.
It has become extra flaky lately, and is preventing people from
using the CQ.

BUG=webrtc:4958

Review URL: https://codereview.webrtc.org/1368763002

Cr-Commit-Position: refs/heads/master@{#10080}
2015-09-25 18:00:54 +00:00
Jiayang Liu
4d47aa335c Fallback to system log when webrtc tracing not enabled.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1368053002 .

Cr-Commit-Position: refs/heads/master@{#10079}
2015-09-25 17:04:36 +00:00
Peter Boström
1741770742 Implement a high-QP threshold for Android H.264.
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.

BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1364253002 .

Cr-Commit-Position: refs/heads/master@{#10078}
2015-09-25 15:03:37 +00:00
henrika
a323fd66de Removes Nexus 6 from OpenSL ES blacklist.
BUG=b/1370703002
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1370703002 .

Cr-Commit-Position: refs/heads/master@{#10077}
2015-09-25 14:25:40 +00:00
tfarina
702f39726b GN: Do not use forward_dependent_configs_from variable.
It is deprecated and public_deps should be used instead, which will have
the
same effect.

BUG=None
R=brettw@chromium.org,kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1365063004

Cr-Commit-Position: refs/heads/master@{#10074}
2015-09-25 12:57:47 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00
henrika
82e20554cb Modifies invalid DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
Ensures that we can restart audio recording on Android without hitting
a DCHECK. Also adds a symmetric design for the playout side.

BUG=webrtc:5000
TEST=modules_unittests --gtest_filter=AudioDevice*

Review URL: https://codereview.webrtc.org/1373443003

Cr-Commit-Position: refs/heads/master@{#10072}
2015-09-25 11:26:19 +00:00
solenberg
3fd7be4cb1 Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}

TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368933002

Cr-Commit-Position: refs/heads/master@{#10069}
2015-09-25 08:36:11 +00:00
solenberg
a53e383d7d Revert of CodecOwner: Don't look at definitions for classes we don't link with (patchset #1 id:1 of https://codereview.webrtc.org/1364233002/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> CodecOwner: Don't look at definitions for classes we don't link with
>
> It's good hygiene and just generally the right thing to do. And
> apparently at least sometimes required by Microsoft's compiler.
>
> Committed: https://crrev.com/f4d38ea57aa739b525066b095468cb4af1d2799b
> Cr-Commit-Position: refs/heads/master@{#10060}

TBR=henrik.lundin@webrtc.org,tommi@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1368083002

Cr-Commit-Position: refs/heads/master@{#10068}
2015-09-25 08:31:01 +00:00
Magnus Jedvert
67e0cf15d3 Android AppRTCDemo: Add slider for changing camera capture quality during call
This CL adds a slider that can change capture resolution and fps during a call. The camera will no be reconfigured, but the frames will be downscaled/dropped in software by cricket::VideoAdapter in the cricket::VideoCapturer. This is controlled with VideoCapturerAndroid.onOutputFormatRequest(). The slider is turned off by default and can be enabled with a checkbox under 'WebRTC Video Settings'.

R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1361083002 .

Cr-Commit-Position: refs/heads/master@{#10067}
2015-09-25 06:23:49 +00:00
kwiberg
574d5daa6d CodecOwner::SetEncoders: Return error code when given bad arguments
Instead of FATAL on a bad codec specification, log and return an error
code. This is a band-aid until callers are taught to only give it good
specifications.

BUG=webrtc:5033, chromium:526478

Review URL: https://codereview.webrtc.org/1364193002

Cr-Commit-Position: refs/heads/master@{#10066}
2015-09-25 05:54:00 +00:00
Jiayang Liu
4ba059d218 Remove custom handler since the logger already logs to console by default.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1367523009 .

Cr-Commit-Position: refs/heads/master@{#10064}
2015-09-24 21:36:17 +00:00
honghaiz
8937437872 Do not prune if the current best connection is weak.
Otherwise, we may delete a useful connection because the current best connection may be failing.

BUG=

Review URL: https://codereview.webrtc.org/1364683002

Cr-Commit-Position: refs/heads/master@{#10063}
2015-09-24 20:14:51 +00:00
deadbeef
59e72ab49b Enable logging for Mac by default on debug builds.
Was previously using the wrong preprocessor define (DEBUG vs _DEBUG).

Review URL: https://codereview.webrtc.org/1361173002

Cr-Commit-Position: refs/heads/master@{#10061}
2015-09-24 17:42:47 +00:00
kwiberg
f4d38ea57a CodecOwner: Don't look at definitions for classes we don't link with
It's good hygiene and just generally the right thing to do. And
apparently at least sometimes required by Microsoft's compiler.

Review URL: https://codereview.webrtc.org/1364233002

Cr-Commit-Position: refs/heads/master@{#10060}
2015-09-24 17:21:02 +00:00
honghaiz
a58ea7806a 1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any.
This will paves the way for continuous nomination lite and multi-networking.
2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.

Review URL: https://codereview.webrtc.org/1311433009

Cr-Commit-Position: refs/heads/master@{#10057}
2015-09-24 15:13:45 +00:00
henrika
8a88dd271e Stability improvement for audio recording on Android
BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1363323002 .

Cr-Commit-Position: refs/heads/master@{#10056}
2015-09-24 14:45:14 +00:00
Magnus Jedvert
7076729c57 Enable SurfaceViewRenderer for AppRTCDemo
BUG=webrtc:4742,webrtc:4910,webrtc:4909
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1356603004 .

Cr-Commit-Position: refs/heads/master@{#10054}
2015-09-24 14:02:15 +00:00
henrika
9236bb1e08 Minor fix for improving logging of supported platform effects
BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1370443003 .

Cr-Commit-Position: refs/heads/master@{#10053}
2015-09-24 13:58:46 +00:00
Erik Språng
6b8d355168 Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/

The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc

BUG=webrtc:4173
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1362303002 .

Cr-Commit-Position: refs/heads/master@{#10052}
2015-09-24 13:07:17 +00:00
Peter Boström
8c266e6baf H264 bitstream parser.
Parsing the encoded bitstream is required for doing downscaling
decisions based on average encoded QP to improve perceived quality.

BUG=webrtc:4968
R=noahric@chromium.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1314473008 .

Cr-Commit-Position: refs/heads/master@{#10051}
2015-09-24 13:07:04 +00:00
kwiberg
ec249d4eae ACMCodecDB: Remove unused stuff, and move private stuff to anonymous namespace
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1360123002

Cr-Commit-Position: refs/heads/master@{#10048}
2015-09-24 11:32:11 +00:00
kwiberg
f66a925142 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1349393003

Cr-Commit-Position: refs/heads/master@{#10046}
2015-09-24 10:18:48 +00:00
will.newton
c675ddd830 video_capture: Better support for UYVY
A couple of places were missing handling of UYVY video formats.

BUG=webrtc:4816

Review URL: https://codereview.webrtc.org/1317613003

Cr-Commit-Position: refs/heads/master@{#10044}
2015-09-24 08:11:45 +00:00
asapersson
74d85e19ae Reduce locking in overuse frame detector now that (as of r9508) the observer_ and options_ can only be set at construction time. E.g. no lock is any longer held while doing the callback.
BUG=

Review URL: https://codereview.webrtc.org/1228853004

Cr-Commit-Position: refs/heads/master@{#10043}
2015-09-24 07:53:38 +00:00
Guo-wei Shieh
53eee43e78 Address the comment from 1367553002.
Remove duplication introduced by
https://codereview.webrtc.org/1367553002

BUG=webrtc:5030
TBR=juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1360203003 .

Cr-Commit-Position: refs/heads/master@{#10039}
2015-09-23 21:09:18 +00:00
Guo-wei Shieh
2e4b620471 TcpPort doesn't connect when calling gmail with non-proxied UDP disabled.
The same check has been made into turnport.cc but missed this place.

BUG=webrtc:5030
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1367553002 .

Cr-Commit-Position: refs/heads/master@{#10038}
2015-09-23 20:57:17 +00:00
Alejandro Luebs
cdfe20bfc1 Fix the maximum native sample rate in AudioProcessing
BUG=webrtc:4983
R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1338833002 .

Cr-Commit-Position: refs/heads/master@{#10037}
2015-09-23 19:49:21 +00:00
deadbeef
cbecd358e0 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/

Which fixes the FYI bots for the original CL, and breaks them for this revert.

Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}

TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1361773005

Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 18:50:31 +00:00
Fredrik Solenberg
d0b5b091e4 Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc.
BUG=webrtc:4690
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1359983003 .

Cr-Commit-Position: refs/heads/master@{#10035}
2015-09-23 15:15:29 +00:00
Peter Boström
7cf0445262 Remove ViEChannel::StartSend deadlock suppression.
No longer lock-order inverting since RTP/RTCP modules are instantiated
on construction and no longer guarded by a separate lock.

BUG=webrtc:2999
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1347283004 .

Cr-Commit-Position: refs/heads/master@{#10034}
2015-09-23 15:04:12 +00:00
Stefan Holmer
8bffba7107 Fix BWE bug where audio has timestamps in us.
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (68786d2040).

BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1360913004 .

Cr-Commit-Position: refs/heads/master@{#10032}
2015-09-23 13:54:04 +00:00
minyuel
6d92bf59f3 Returning correct duration estimate on Opus DTX packets.
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,

2. NetEq DoCodecInternalCng did not assign enough buffer.

P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.

BUG=webrtc:4985
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1334303005 .

Cr-Commit-Position: refs/heads/master@{#10031}
2015-09-23 13:20:56 +00:00
henrika
c14f5ff60f Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.

R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE

Review URL: https://codereview.webrtc.org/1344563002 .

Cr-Commit-Position: refs/heads/master@{#10030}
2015-09-23 12:09:40 +00:00
Erik Språng
c9bbeb0354 Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29

Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}

TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1362923002 .

Cr-Commit-Position: refs/heads/master@{#10029}
2015-09-23 11:52:01 +00:00
Peter Boström
d5c75b1a0b Reduce LS_INFO spam from voice_engine/.
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).

BUG=
R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1347353004 .

Cr-Commit-Position: refs/heads/master@{#10028}
2015-09-23 11:24:43 +00:00
torbjorng
a81a42f584 Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.

Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1358413003

Cr-Commit-Position: refs/heads/master@{#10024}
2015-09-23 09:24:27 +00:00
ivica
2d4e6c5d9d Fixing camera capture for video_loopback
In the middle of refactoring, I replaced the VideoCapturer with
FrameGeneratorCapturer, to reuse the code, and with that disabled the camera.
Now adding capturer_ element to VideoQualityTest and ignoring
frame_generator_capturer_ from the parent class test::CallTest.

Review URL: https://codereview.webrtc.org/1356933005

Cr-Commit-Position: refs/heads/master@{#10023}
2015-09-23 08:57:13 +00:00
deadbeef
47ee2f3b9f TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

Review URL: https://codereview.webrtc.org/1350523003

Cr-Commit-Position: refs/heads/master@{#10022}
2015-09-22 22:08:31 +00:00
kwiberg
8967183bf7 Simple cleanups of AudioDecoder and AudioEncoder classes
* Make sure they're all final and don't allow copying or assignment.

  * Get rid of the single-channel PCM decoder classes.

  * Move some includes from .h to .cc files where possible.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1353803002

Cr-Commit-Position: refs/heads/master@{#10021}
2015-09-22 21:06:34 +00:00
torbjorng
07d09364b0 Purge nss files and dependencies.
This replaces https://codereview.webrtc.org/1313233005
which was reverted after triggering Chromium issues.
The only difference is that we're cleaned up dependencies
on use_openssl from the gyp file.

Since https://codereview.chromium.org/1358913003 landed,
this CL should cause no Chromium issues.

BUG=webrtc:4497

Review URL: https://codereview.webrtc.org/1351503004

Cr-Commit-Position: refs/heads/master@{#10019}
2015-09-22 18:58:13 +00:00
Karl Wiberg
7404368998 Move AudioDecoderIsac* to its own files
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1339253003 .

Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22 17:31:52 +00:00
Peter Boström
7083e119e8 Remove callback_cs_ in ViEEncoder.
Instead make callbacks const and set on construction.

BUG=webrtc:1695
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/1354143004 .

Cr-Commit-Position: refs/heads/master@{#10017}
2015-09-22 14:29:00 +00:00
kwiberg
6faf5bebba Move AudioDecoderPcm* next to AudioEncoderPcm*
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1348613003

Cr-Commit-Position: refs/heads/master@{#10015}
2015-09-22 13:16:56 +00:00
ivica
d4818e7304 Using static frame generator when no scrolling
In screensharing full stack tests, instead of using YuvFileGenerator by default
when no scrolling is used, I always used ScrollingImageFileGenerator.
That possibly slowed down the test a little bit, at least for the slowed
devices, as it unnecessarily copied few MBs per frame.

BUG=chromium:534220

Review URL: https://codereview.webrtc.org/1359783002

Cr-Commit-Position: refs/heads/master@{#10014}
2015-09-22 12:47:34 +00:00
Henrik Boström
9b5476de9a sslidentity.cc/IntKeyTypeFamilyToKeyType function added, converting from int to KeyType.
Added to prevent Chromium from breaking if KeyType (now an enum) starts being used in Chromium before KeyType changes to a parameterizable class. When enum -> class change happens, IntKeyTypeFamilyToKeyType will be updated at the same time.

Once Chromium starts using class KeyType with parameters this function can be removed.

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1363543002 .

Cr-Commit-Position: refs/heads/master@{#10013}
2015-09-22 12:13:23 +00:00
sprang
ef165eefc7 Wire up send-side bandwidth estimation.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1338203003

Cr-Commit-Position: refs/heads/master@{#10012}
2015-09-22 12:10:58 +00:00