13696 Commits

Author SHA1 Message Date
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
Taylor Brandstetter
fe69a74ba8 Making ContinueSSL synchronously set the state to "open".
It was recently made asynchronous, and this broke some downstream tests.

BUG=webrtc:6387
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2381313003 .

Cr-Commit-Position: refs/heads/master@{#14463}
2016-10-01 00:34:42 +00:00
deadbeef
dd7fb43f28 Emit SignalReadyToSend even for "presumed writable" connections.
The Connection class will now blindly forward SignalReadyToSend, and
P2PTransportChannel will decide whether to forward it further (which
it was already doing).

BUG=webrtc:6448

Review-Url: https://codereview.webrtc.org/2374183005
Cr-Commit-Position: refs/heads/master@{#14462}
2016-09-30 22:16:57 +00:00
deadbeef
89824f6fe0 Relanding: Allow the DTLS fingerprint verification to occur after the handshake.
This means the DTLS handshake can make progress while the SDP answer
containing the fingerprint is still in transit. If the signaling path
if significantly slower than the media path, this can have a moderate
impact on call setup time.

Of course, until the fingerprint is verified no media can be sent. Any
attempted write will result in SR_BLOCK.

This essentially fulfills the requirements of RFC 4572, Section 6.2:

   Note that when the offer/answer model is being used, it is possible
   for a media connection to outrace the answer back to the offerer.
   Thus, if the offerer has offered a 'setup:passive' or 'setup:actpass'
   role, it MUST (as specified in RFC 4145 [2]) begin listening for an
   incoming connection as soon as it sends its offer.  However, it MUST
   NOT assume that the data transmitted over the TLS connection is valid
   until it has received a matching fingerprint in an SDP answer.  If
   the fingerprint, once it arrives, does not match the client's
   certificate, the server endpoint MUST terminate the media connection
   with a bad_certificate error, as stated in the previous paragraph.

BUG=webrtc:6387

Review-Url: https://codereview.webrtc.org/2163683003
Cr-Commit-Position: refs/heads/master@{#14461}
2016-09-30 18:55:49 +00:00
stefan
3cdfcd88a1 Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )
Reason for revert:
Broke browser_tests, e.g., WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264

Original issue's description:
> Use sps and pps to determine decodability of H.264 frames.
>
> NOPRESUBMIT=true
> BUG=webrtc:6208
> R=philipel@webrtc.org
>
> Committed: https://crrev.com/17b02633666f2f5d7e78767ad5674c728d639c26
> Cr-Commit-Position: refs/heads/master@{#14458}

TBR=philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2381233004
Cr-Commit-Position: refs/heads/master@{#14460}
2016-09-30 16:06:43 +00:00
gaetano.carlucci
61050f67ef Fixig issues in BWE dynamics plot scripts.
BUG=None

Review-Url: https://codereview.webrtc.org/2360053003
Cr-Commit-Position: refs/heads/master@{#14459}
2016-09-30 13:29:57 +00:00
Stefan Holmer
17b0263366 Use sps and pps to determine decodability of H.264 frames.
NOPRESUBMIT=true
BUG=webrtc:6208
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2341713002 .

Cr-Commit-Position: refs/heads/master@{#14458}
2016-09-30 13:24:26 +00:00
sakal
55d932b331 Add logging statements to places where the frame might be dropped in WebRTC pipeline.
BUG=b/31645554

Review-Url: https://codereview.webrtc.org/2361803003
Cr-Commit-Position: refs/heads/master@{#14457}
2016-09-30 13:19:12 +00:00
nisse
115bd153c7 New helper function test::ReadI420Buffer, refactor FrameReader to use it.
This change reduces the number of places where we first fread a I420
frame into a uint8_t buffer, followed by a copy into a frame buffer
object.

BUG=None

Review-Url: https://codereview.webrtc.org/2362683002
Cr-Commit-Position: refs/heads/master@{#14456}
2016-09-30 11:14:11 +00:00
nisse
6f112cc136 Delete unused support for vp8 partitions.
This also makes it possible to drop the RTPFragmentationHeader from
the class VCMEncodedFrame.

BUG=None

Review-Url: https://codereview.webrtc.org/2380933003
Cr-Commit-Position: refs/heads/master@{#14455}
2016-09-30 10:43:07 +00:00
asapersson
3cc47ebd2d Add sanity check for decreasing RTP timestamp in RtpToNtpMs.
The capture time for a frame (capture_ms) is set later (in ViEEncoder::IncomingCapturedFrame) than the timestamp.
Could potentially cause the RTP timestamp in consecutive RTCP SR to decrease.
Example:
// Frame1 46371: timestamp:2732, capture_ms:46373, rtcp SR ms: 46423 -> estimated current RTP timestamp:2732+(46423-46373)*90 = 7232
// Frame2 46404: timestamp:5702, capture_ms:46412, rtcp SR ms: 46428 -> estimated current RTP timestamp:5702+(46428-46412)*90 = 7142
// Diff:  33 ms:          33 ms,            39 ms,              5 ms

BUG=b/31154867

Review-Url: https://codereview.webrtc.org/2354843003
Cr-Commit-Position: refs/heads/master@{#14454}
2016-09-30 10:16:26 +00:00
nisse
f5297a019e Reland of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #1 id:1 of https://codereview.webrtc.org/2357113002/ )
Reason for revert:
Downstream code is being fixed.

Original issue's description:
> Revert of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #9 id:160001 of https://codereview.webrtc.org/2262443003/ )
>
> Reason for revert:
> Breaks downstream testcode, still using CapturedFrame.
>
> Original issue's description:
> > Delete VideoFrameFactory, CapturedFrame, and related code.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/66ac50e58c790624d51ede10ae438cbadbca9d2e
> > Cr-Commit-Position: refs/heads/master@{#14315}
>
> TBR=pthatcher@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/66492210e57ee8efce2ad4d45a8781df1fcaa5e3
> Cr-Commit-Position: refs/heads/master@{#14320}

TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2370993003
Cr-Commit-Position: refs/heads/master@{#14453}
2016-09-30 08:34:35 +00:00
Stefan Holmer
280de9e1c3 Reland: Fix race / crash in OnNetworkRouteChanged().
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
  result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
  unnecessary dependencies.

Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.

BUG=webrtc:6427, webrtc:6422
R=terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2378103005 .

Cr-Commit-Position: refs/heads/master@{#14452}
2016-09-30 08:07:00 +00:00
kthelgason
20a52e1639 Reland of Unify the macOS and iOS capturer implementations (patchset #1 id:1 of https://codereview.webrtc.org/2381853002/ )
Reason for revert:
Internal project has been fixed

Original issue's description:
> Revert of Unify the macOS and iOS capturer implementations (patchset #4 id:60001 of https://codereview.webrtc.org/2309253005/ )
>
> Reason for revert:
> Breaks internal project
>
> Original issue's description:
> > Unify the macOS and iOS capturer implementations
> >
> > This removes the QTKit based capturer for mac, and removes the need
> > to link against deprecated system libraries on macOS.
> >
> > BUG=webrtc:3968,webrtc:6275,webrtc:6333
> >
> > Committed: https://crrev.com/242d8bdddd77109781cbb70c59d161be7566ac98
> > Cr-Commit-Position: refs/heads/master@{#14418}
>
> TBR=magjed@webrtc.org,tkchin@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:3968,webrtc:6275,webrtc:6333
>
> Committed: https://crrev.com/eddb7571d81e51a66f4abaf55013c85b4132c837
> Cr-Commit-Position: refs/heads/master@{#14429}

TBR=magjed@webrtc.org,tkchin@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:3968,webrtc:6275,webrtc:6333

Review-Url: https://codereview.webrtc.org/2375273004
Cr-Commit-Position: refs/heads/master@{#14451}
2016-09-30 07:43:18 +00:00
denicija
edbae5e0ac Remove Crit::Scope lock by using atomic bool property.
The clang static analyzer seems unable to resolve cpp locks in ObjC code.
As of current time, the clang analyzer has known limitations documented
http://clang.llvm.org/docs/ThreadSafetyAnalysis.html#known-limitations.
From the documentation: "The analysis currently does not do any checking
inside constructors or destructors.
In other words, every constructor and destructor is treated
as if it was annotated with NO_THREAD_SAFETY_ANALYSIS."
This is 'probably' why the analyzer is unable to resolve the lock when
used in ObjC land (the cpp works fine).
The lock can be removed by using atomic property instead.
It's not on performance critical path and we expect updates on just one queue and reads from others. That's why the thread assurance atomic properties bring is enough.
The CL removes rtc_sdk_peerconnection_objc_warnings_config as well as it's no longer needed.

BUG=webrtc:6308

Review-Url: https://codereview.webrtc.org/2372513004
Cr-Commit-Position: refs/heads/master@{#14450}
2016-09-30 07:21:28 +00:00
ehmaldonado
eb5040ae44 Disable TCPChannelClientTest.testConnectIPv6
This test is failing because IPv6 is not fully supported on some of the
bots. (See https://bugs.chromium.org/p/chromium/issues/detail?id=612380)

R=kjellander@webrtc.org
TBR=perkj@webrtc.org
BUG=webrtc:6437
NOTRY=True
TESTED=ninja -C out/gn && out/gn/bin/run_android_junit_tests with and without the CL and verified the test is not run.

Review-Url: https://codereview.webrtc.org/2381503005
Cr-Commit-Position: refs/heads/master@{#14449}
2016-09-30 07:20:12 +00:00
Kári Tristan Helgason
15e4ec334c Remove compat for iOS 7/8
BUG=None
R=magjed@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/2382713002 .

Cr-Commit-Position: refs/heads/master@{#14448}
2016-09-30 06:56:44 +00:00
perkj
3b703ede8b Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
Reason for revert:
Fails on a content_browsertest (and also webrtc_perf?)

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/34336

https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/9091/steps/webrtc_perf_tests/logs/stdio
[  FAILED  ] FullStackTest.ParisQcifWithoutPacketLoss (59436 ms)

Original issue's description:
> Let ViEEncoder handle resolution changes.
>
> This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
>
> With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
>
> BUG=webrtc:5687, webrtc:6371, webrtc:5332
>
> Committed: https://crrev.com/26105b41b4f97642ee30cb067dc786c2737709ad
> Cr-Commit-Position: refs/heads/master@{#14445}

TBR=sprang@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2383493005
Cr-Commit-Position: refs/heads/master@{#14447}
2016-09-30 06:25:46 +00:00
Honghai Zhang
b73d269707 Replace RelayPort with TurnPort in p2ptransportchannel tests.
Also remove the relay servers in the tests.
Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.

BUG=None
R=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/2380923002 .

Committed: https://crrev.com/c8d21712dde64c7d613d1ea56c840438505a909f
Cr-Original-Commit-Position: refs/heads/master@{#14441}
Cr-Commit-Position: refs/heads/master@{#14446}
2016-09-30 05:46:16 +00:00
perkj
26105b41b4 Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
2016-09-30 05:39:15 +00:00
honghaiz
8b8459e31b suppress memcheck test errors in the Opus decoder and encoder.
BUG=webrtc:6444

TBR=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2374253006
Cr-Commit-Position: refs/heads/master@{#14444}
2016-09-30 04:12:46 +00:00
honghaiz
75f6626718 Revert of Replace RelayPort with TurnPort in p2ptransportchannel tests. (patchset #2 id:40001 of https://codereview.webrtc.org/2380923002/ )
Reason for revert:
It caused some tests in p2ptransportchannel flaky.

Original issue's description:
> Replace RelayPort with TurnPort in p2ptransportchannel tests.
>
> Also remove the relay servers in the tests.
> Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.
>
> BUG=None
> R=deadbeef@webrtc.org
>
> Committed: https://crrev.com/c8d21712dde64c7d613d1ea56c840438505a909f
> Cr-Commit-Position: refs/heads/master@{#14441}

TBR=deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2385563002
Cr-Commit-Position: refs/heads/master@{#14443}
2016-09-30 01:17:36 +00:00
danilchap
7851bda9bc Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
move all logic from that class into RTCPReceiver too,
Simplify and fix style on the way.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2373053002
Cr-Commit-Position: refs/heads/master@{#14442}
2016-09-29 22:28:12 +00:00
Honghai Zhang
c8d21712dd Replace RelayPort with TurnPort in p2ptransportchannel tests.
Also remove the relay servers in the tests.
Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.

BUG=None
R=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/2380923002 .

Cr-Commit-Position: refs/heads/master@{#14441}
2016-09-29 21:51:36 +00:00
skvlad
7502401788 Do not spam "Connect failed with 101/65" in logs.
WebRTC polls the list of local IP addresses for both IPv4 and IPv6
every ~2 seconds. It does so by trying to connect() a UDP socket to
an address on the public Internet (without actually sending any
packets).

If the host doesn't have IPv6 (or IPv4) connectivity, it fails with
errno 101 (ENETUNREACH, Linux) or errno 65 (EHOSTUNREACH, Mac).

This is the expected behavior, and we shouldn't be logging these
failures, especially since polling is fairly frequent.

BUG=webrtc:6347
R=deadbeef@webrtc.org, honghaiz@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/2370383002 .

Cr-Commit-Position: refs/heads/master@{#14440}
2016-09-29 19:59:44 +00:00
henrik.lundin
591c709fcb Suppress a memcheck error in Opus decoder
BUG=webrtc:6444
TBR=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2376103004
Cr-Commit-Position: refs/heads/master@{#14439}
2016-09-29 15:45:26 +00:00
phoglund
590cf281fb Add autothread to pseudo-tcp fuzzer.
I think this will make a rtc::Thread object exist for the lifetime of
the environment, which will remove some uninteresting crashes.

BUG=chrome:648075

Review-Url: https://codereview.webrtc.org/2365373002
Cr-Commit-Position: refs/heads/master@{#14438}
2016-09-29 13:27:52 +00:00
sakal
70736e4c9d Remove old presumably unused directory.
BUG=None

Review-Url: https://codereview.webrtc.org/2378103002
Cr-Commit-Position: refs/heads/master@{#14437}
2016-09-29 12:38:06 +00:00
isheriff
8e6a76194d ProbeController: Limit max probing bitrate
BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2352093002
Cr-Commit-Position: refs/heads/master@{#14436}
2016-09-29 12:37:05 +00:00
magjed
606018600e Add presubmit format requirement for webrtc/api/android
BUG=webrtc:6419
NOTRY=True
TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2377113003
Cr-Commit-Position: refs/heads/master@{#14435}
2016-09-29 12:36:03 +00:00
Henrik Lundin
56145667b8 Fix faulty include paths that break the build
BUG=none
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2380013002 .

Cr-Commit-Position: refs/heads/master@{#14434}
2016-09-29 12:34:39 +00:00
stefan
5ec85fbcb7 Revert of Fix race / crash in OnNetworkRouteChanged(). (patchset #5 id:80001 of https://codereview.webrtc.org/2366333003/ )
Reason for revert:
Caused issues with webrtc_perf_tests on build bots.

Original issue's description:
> Fix race / crash in OnNetworkRouteChanged().
>
> To achieve this some refactoring was done to make it possible to synchronize
> access to the DelayBasedBwe in TransportFeedbackAdapter:
> - The callback was removed from DelayBasedBwe, it now instead returns its
>   result.
> - TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
>   unnecessary dependencies.
>
> Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
>
> BUG=webrtc:6427, webrtc:6422
>
> Committed: https://crrev.com/fd0d42669204e6dd92a60736bca7ae0196663024
> Cr-Commit-Position: refs/heads/master@{#14430}

TBR=terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6427, webrtc:6422

Review-Url: https://codereview.webrtc.org/2377303002
Cr-Commit-Position: refs/heads/master@{#14433}
2016-09-29 11:19:42 +00:00
sakal
b6760f9e44 Format all Java in WebRTC.
BUG=webrtc:6419
TBR=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2377003002
Cr-Commit-Position: refs/heads/master@{#14432}
2016-09-29 11:12:51 +00:00
Per
a48ddb7636 Add VideoSendStream::Stats::prefered_media_bitrate_bps
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.

BUG=webrtc:6371
R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2368223002 .

Cr-Commit-Position: refs/heads/master@{#14431}
2016-09-29 09:49:01 +00:00
stefan
fd0d426692 Fix race / crash in OnNetworkRouteChanged().
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
  result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
  unnecessary dependencies.

Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.

BUG=webrtc:6427, webrtc:6422

Review-Url: https://codereview.webrtc.org/2366333003
Cr-Commit-Position: refs/heads/master@{#14430}
2016-09-29 09:44:38 +00:00
kthelgason
eddb7571d8 Revert of Unify the macOS and iOS capturer implementations (patchset #4 id:60001 of https://codereview.webrtc.org/2309253005/ )
Reason for revert:
Breaks internal project

Original issue's description:
> Unify the macOS and iOS capturer implementations
>
> This removes the QTKit based capturer for mac, and removes the need
> to link against deprecated system libraries on macOS.
>
> BUG=webrtc:3968,webrtc:6275,webrtc:6333
>
> Committed: https://crrev.com/242d8bdddd77109781cbb70c59d161be7566ac98
> Cr-Commit-Position: refs/heads/master@{#14418}

TBR=magjed@webrtc.org,tkchin@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:3968,webrtc:6275,webrtc:6333

Review-Url: https://codereview.webrtc.org/2381853002
Cr-Commit-Position: refs/heads/master@{#14429}
2016-09-29 09:43:27 +00:00
magjed
ff9793c600 Android: Remove onOutputFormatRequest from the VideoCapturer interface
Remove onOutputFormatRequest from the VideoCapturer interface and from
all implementations of that interface. Apps should now use
VideoSource.adaptOutputFormat() instead.

BUG=webrtc:6391

Review-Url: https://codereview.webrtc.org/2373353002
Cr-Commit-Position: refs/heads/master@{#14428}
2016-09-29 09:14:39 +00:00
isheriff
90ce01dbbe The current default schedule delay of 30 ms prohibits
scaling to high bitrates when probing.

    BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2372013002
Cr-Commit-Position: refs/heads/master@{#14427}
2016-09-29 09:02:16 +00:00
johan
0fd22ef0ae Rename P2PTransportChannel worker_thread_ to network_thread_.
Restore consistency of thread names in ThreadController and P2PTransportChannel.
This is a follow-up for https://codereview.webrtc.org/1895813003 and https://codereview.webrtc.org/1903393004.

BUG=webrtc:6432

Review-Url: https://codereview.webrtc.org/2378573003
Cr-Commit-Position: refs/heads/master@{#14426}
2016-09-29 08:19:28 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
mattdr
51f29197e6 Update WebRTC to build against libsrtp 2.0
BUG=webrtc:6376

Review-Url: https://codereview.webrtc.org/2345753002
Cr-Commit-Position: refs/heads/master@{#14424}
2016-09-28 21:08:53 +00:00
kwiberg
24c7c1238d Move FunctionView from AudioCodingModule to the rtc namespace
It's a very general type, and we're about to start needing it in other
places besides AudioCodingModule.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2380463003
Cr-Commit-Position: refs/heads/master@{#14423}
2016-09-28 18:57:17 +00:00
ehmaldonado
35d43b90b9 Roll chromium_revision bdaa23ddfe..316b880c55 (421490:421519)
Change log: bdaa23ddfe..316b880c55
Full diff: bdaa23ddfe..316b880c55

Changed dependencies:
* src/third_party/catapult: 03fbd54d8f..f332dd6dd5
DEPS diff: bdaa23ddfe..316b880c55/DEPS

Clang version changed 282097:282487
Details: bdaa23ddfe..316b880c55/tools/clang/scripts/update.py

TBR=kjellander@webrtc.org
NOTRY=True
BUG=None

Review-Url: https://codereview.webrtc.org/2379603002
Cr-Commit-Position: refs/heads/master@{#14422}
2016-09-28 17:55:15 +00:00
deadbeef
7e146cb97e Fixing heap read overflow when "sctp-port" is in a video description.
This added an SCTP codec, which is later re-interpreted as a video
codec. We shouldn't be adding codecs that don't match the type of the
media description.

BUG=chromium:648062

Review-Url: https://codereview.webrtc.org/2354723002
Cr-Commit-Position: refs/heads/master@{#14421}
2016-09-28 17:04:41 +00:00
kthelgason
478681e1e6 Move the QP scaling thresholds to the relevant encoders.
Also provide a new set of thresholds for the VideoToolbox encoder. The new thresholds were experimentally determined to work well on the iPhone 6S, and also adequately on the iPhone 5S.

BUG=webrtc:5678

Review-Url: https://codereview.webrtc.org/2309743002
Cr-Commit-Position: refs/heads/master@{#14420}
2016-09-28 15:17:51 +00:00
palmkvist
e75f204b06 Expose Ivf logging through the native API
BUG=webrtc:6300

Review-Url: https://codereview.webrtc.org/2303273002
Cr-Commit-Position: refs/heads/master@{#14419}
2016-09-28 13:19:53 +00:00
kthelgason
242d8bdddd Unify the macOS and iOS capturer implementations
This removes the QTKit based capturer for mac, and removes the need
to link against deprecated system libraries on macOS.

BUG=webrtc:3968,webrtc:6275,webrtc:6333

Review-Url: https://codereview.webrtc.org/2309253005
Cr-Commit-Position: refs/heads/master@{#14418}
2016-09-28 12:51:44 +00:00
buildbot
f5e3bbe3d6 Roll chromium_revision 386676ff4e..bdaa23ddfe (421470:421490)
Change log: 386676ff4e..bdaa23ddfe
Full diff: 386676ff4e..bdaa23ddfe

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2371383004
Cr-Commit-Position: refs/heads/master@{#14417}
2016-09-28 11:44:14 +00:00
nisse
e5684c5387 Delete method webrtc::VideoFrame::allocated_size and enum PlaneType.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2380623002
Cr-Commit-Position: refs/heads/master@{#14416}
2016-09-28 10:14:15 +00:00
danilchap
798896a4aa Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
structs are exactly the same but last one follow naming style.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2368983002
Cr-Commit-Position: refs/heads/master@{#14415}
2016-09-28 09:54:30 +00:00