Commit Graph

16500 Commits

Author SHA1 Message Date
1bed2e486e video_loopback: fall back to fake capturer if we can't open camera.
Test manually, since it's a manual test.

BUG=webrtc:7036

Review-Url: https://codereview.webrtc.org/2652713002
Cr-Commit-Position: refs/heads/master@{#16218}
2017-01-23 16:46:51 +00:00
435ddf978d Add TransportFeedbackPacketLossTracker.
This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder.

BUG=webrtc:6904

Review-Url: https://codereview.webrtc.org/2579613003
Cr-Commit-Position: refs/heads/master@{#16217}
2017-01-23 16:07:05 +00:00
ed582f7e36 Script to start stubbed loopback video test with Espresso
BUG=webrtc:7034

Review-Url: https://codereview.webrtc.org/2632323003
Cr-Commit-Position: refs/heads/master@{#16216}
2017-01-23 15:55:42 +00:00
0ebdf2757c Delete or update left-over ASSERT use and comments.
BUG=webrtc:6424,webrtc:6323

Review-Url: https://codereview.webrtc.org/2647663002
Cr-Commit-Position: refs/heads/master@{#16215}
2017-01-23 15:43:05 +00:00
da25006431 Fixed public_deps for libjingle_peerconnection{,_api}
https://codereview.webrtc.org/2514883002/ changed and moved these targets around but did not add public dependencies for the fallbacks, which causes gn gen --check a lot of anger.

NOTRY=true # Only build changes and windows bots are cranky atm.
BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2651663002
Cr-Commit-Position: refs/heads/master@{#16214}
2017-01-23 15:37:43 +00:00
50cfe1fda7 RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-framesdropped
Implemented as frames_received - frames_rendered.

Part of this CL is adding frames_rendered to VideoReceiveStream::Stats
and updating it at ReceiveStatisticsProxy::OnRenderedFrame.

BUG=webrtc:6757, chromium:659137, chromium:627816
NOTRY=True

Review-Url: https://codereview.webrtc.org/2607933002
Cr-Commit-Position: refs/heads/master@{#16213}
2017-01-23 15:21:55 +00:00
9c3d4c4d88 Stop leaking FlexfecReceiveStream objects after call shutdown.
BUG=webrtc:7017

Review-Url: https://codereview.webrtc.org/2645703003
Cr-Commit-Position: refs/heads/master@{#16212}
2017-01-23 14:59:13 +00:00
a067013e90 Minor style change suggested by internal static analysis tool.
BUG=None

Review-Url: https://codereview.webrtc.org/2645333003
Cr-Commit-Position: refs/heads/master@{#16211}
2017-01-23 14:09:35 +00:00
7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings.
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.

Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.

Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.

BUG=webrtc:5883

Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 12:56:25 +00:00
f49ff260d1 GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false
BUG=webrtc:6626
NOTRY=True

Review-Url: https://codereview.webrtc.org/2647003002
Cr-Commit-Position: refs/heads/master@{#16209}
2017-01-23 12:26:02 +00:00
fd870db0b2 Add metric for decode time and max decode time in video quality tests.
BUG=chromium:672007

Review-Url: https://codereview.webrtc.org/2640263002
Cr-Commit-Position: refs/heads/master@{#16208}
2017-01-23 11:22:15 +00:00
011240333e Minor style change suggested by internal static analysis tool.
TBR=sakal@webrtc.org
BUG=None

Review-Url: https://codereview.webrtc.org/2646413002
Cr-Commit-Position: refs/heads/master@{#16207}
2017-01-23 11:10:39 +00:00
a31cdbce13 Roll chromium_revision dcc5978539..59592eaa98 (445328:445345)
Change log: dcc5978539..59592eaa98
Full diff: dcc5978539..59592eaa98

Changed dependencies:
* src/build: e4091c319b..92cbc7ad5a
* src/third_party: 2ed7d64dc8..56ef664776
DEPS diff: dcc5978539..59592eaa98/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2652633002
Cr-Commit-Position: refs/heads/master@{#16206}
2017-01-23 11:09:34 +00:00
0b56279da1 Catch failure to load native dependencies.
BUG=webrtc:6751

Review-Url: https://codereview.webrtc.org/2652623002
Cr-Commit-Position: refs/heads/master@{#16205}
2017-01-23 11:02:56 +00:00
de8ca92755 New script to count usage of C++ classes.
This script is similar to header_usage.sh, but it counts usage (number
of files) of each class defined in some header file.

E.g., the following classes appear unused,

  AsyncHttpsProxyServerSocket
  CompositeMediaEngineWithFakeVoiceEngine
  DesktopId
  RtpDataCallback
  ScopedMessageData
  ScreencastEventCatcher

NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2649103002
Cr-Commit-Position: refs/heads/master@{#16204}
2017-01-23 10:41:08 +00:00
b55bd97972 Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ )
Reason for revert:
It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places.

Original issue's description:
> Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
>
> Reason for revert:
> This breaks some chromium.webrtc.fyi buildbots with the following error:
>
> ERROR Unresolved dependencies.
> //third_party/webrtc/base:base(//build/toolchain/android:android_arm)
>   needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
>
>
> Original issue's description:
> > Creating libwebrtc bundle jar
> >
> > Creates a JAR which includes:
> > - //webrtc/base:base_java
> > - //webrtc/modules/audio_device:audio_device_java
> > - //webrtc/sdk/android:libjingle_peerconnection_java
> > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
> >
> > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2646443002
> > Cr-Commit-Position: refs/heads/master@{#16189}
> > Committed: a62a82b7e7
>
> TBR=kjellander@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2640023010
> Cr-Commit-Position: refs/heads/master@{#16190}
> Committed: 3c9151b953

TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356

Review-Url: https://codereview.webrtc.org/2646093004
Cr-Commit-Position: refs/heads/master@{#16203}
2017-01-23 09:25:53 +00:00
5d0f2e8753 Roll chromium_revision 269b6bc66e..dcc5978539 (445317:445328)
Change log: 269b6bc66e..dcc5978539
Full diff: 269b6bc66e..dcc5978539

Changed dependencies:
* src/third_party: 501c470ab5..2ed7d64dc8
DEPS diff: 269b6bc66e..dcc5978539/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2648153003
Cr-Commit-Position: refs/heads/master@{#16202}
2017-01-23 07:26:44 +00:00
c1524347e9 Roll chromium_revision 7649e76842..269b6bc66e (445027:445317)
Change log: 7649e76842..269b6bc66e
Full diff: 7649e76842..269b6bc66e

Changed dependencies:
* src/base: 4f9b6b4f4e..d23c26e094
* src/build: 43ba3a29da..e4091c319b
* src/testing: c11b06231a..52fa364615
* src/third_party: e6dcf7494f..501c470ab5
* src/third_party/catapult: 49e3f62b24..e1e778d78d
* src/tools: 5012bd139b..b12278e6ba
DEPS diff: 7649e76842..269b6bc66e/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2649023002
Cr-Commit-Position: refs/heads/master@{#16201}
2017-01-23 05:03:34 +00:00
3e4faae0ed Fixing memory leak in FakeTransportController.
Introduced by: https://codereview.webrtc.org/2641633002/
Only occurs with test code.

BUG=webrtc:6972
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/2648093002
Cr-Commit-Position: refs/heads/master@{#16200}
2017-01-21 06:43:34 +00:00
8662f94023 Only set certificate on DTLS transport if fingerprint is found in SDP.
This is used for fallback from DTLS to SDES encryption, which we probably still
want to support. Setting a certificate puts the DTLS transport in a "DTLS
enabled" mode, so it should be delayed until SDP with "a=fingerprint" is set.

BUG=webrtc:6972

Review-Url: https://codereview.webrtc.org/2641633002
Cr-Commit-Position: refs/heads/master@{#16199}
2017-01-21 05:20:51 +00:00
2197e91860 Remove dead code for GtkVideoRenderer.
Pulls in unnecessary GTK dependencies that breaks the chromium GTK3
build. This removes the last of webrtc/media/devices.

BUG=chromium:668446

Review-Url: https://codereview.webrtc.org/2646793008
Cr-Commit-Position: refs/heads/master@{#16198}
2017-01-21 02:13:07 +00:00
f33491ebaf Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ )
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.

Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: eaa826c2ee

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025

Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
2017-01-21 01:01:45 +00:00
eaa826c2ee Removing #defines previously used for building without BoringSSL/OpenSSL.
These defines don't work any more, so they only cause confusion:

FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL

BUG=webrtc:7025

Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16196}
2017-01-20 23:15:58 +00:00
cd3180c0c6 PATENTS: fix reference
The 'src' path referred to in the PATENTS.txt has been renamed
to 'webrtc'.

BUG=webrtc:7021

Review-Url: https://codereview.webrtc.org/2640143004
Cr-Commit-Position: refs/heads/master@{#16195}
2017-01-20 20:45:07 +00:00
7bcdb69957 Ignore ufrag/password in "a=candidate" lines in SDP.
These attributes should be parsed in candidate trickling, but when
parsing a full session description, only "a=ice-ufrag"/"a=ice-pwd"
should be used to communicate the ufrag/password.

BUG=chromium:681286

Review-Url: https://codereview.webrtc.org/2639183002
Cr-Commit-Position: refs/heads/master@{#16194}
2017-01-20 20:43:58 +00:00
0fc04b74c5 Finalize the support for building WebRTC library for iOS with bitcode
Initial provisioning was already done in build_ios_libs.sh to support
building the WebRTC framework or static library for iOS (tvOS, watchOS)
with bitcode. Still, the actual build configuration would need to be
modified for each and every part of the build, including 3rd-party libs.
Thus, doing that more universally, at the build/config level, would be
desirable – and actually necessary to provide the intended support.

The patch for enhancing the Chromium build configs with that specific
option was landed in https://codereview.chromium.org/2631573002

NOTRY=True
BUG=webrtc:5085

Review-Url: https://codereview.webrtc.org/2633643003
Cr-Commit-Position: refs/heads/master@{#16193}
2017-01-20 16:01:36 +00:00
f64941f1a5 RTCMediaStreamTrackStats.framesDecoded collected.
According to spec:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-framesdecoded

BUG=webrtc:6757, chromium:659137, chromium:627816

Review-Url: https://codereview.webrtc.org/2642713004
Cr-Commit-Position: refs/heads/master@{#16192}
2017-01-20 15:39:09 +00:00
aea1a017ed Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS
The folder webrtc/sdk/ now contains android as well, so we should move the objc DEPS file to the objc folder.

TBR=tommi
BUG=None

Review-Url: https://codereview.webrtc.org/2644733008
Cr-Commit-Position: refs/heads/master@{#16191}
2017-01-20 14:56:44 +00:00
3c9151b953 Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
Reason for revert:
This breaks some chromium.webrtc.fyi buildbots with the following error:

ERROR Unresolved dependencies.
//third_party/webrtc/base:base(//build/toolchain/android:android_arm)
  needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)

Original issue's description:
> Creating libwebrtc bundle jar
>
> Creates a JAR which includes:
> - //webrtc/base:base_java
> - //webrtc/modules/audio_device:audio_device_java
> - //webrtc/sdk/android:libjingle_peerconnection_java
> - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
>
> The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2646443002
> Cr-Commit-Position: refs/heads/master@{#16189}
> Committed: a62a82b7e7

TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356

Review-Url: https://codereview.webrtc.org/2640023010
Cr-Commit-Position: refs/heads/master@{#16190}
2017-01-20 14:48:03 +00:00
a62a82b7e7 Creating libwebrtc bundle jar
Creates a JAR which includes:
- //webrtc/base:base_java
- //webrtc/modules/audio_device:audio_device_java
- //webrtc/sdk/android:libjingle_peerconnection_java
- //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java

The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.

BUG=webrtc:6356

Review-Url: https://codereview.webrtc.org/2646443002
Cr-Commit-Position: refs/heads/master@{#16189}
2017-01-20 14:15:34 +00:00
fefe076789 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector.
BUG=webrtc:6757, chromium:659137, chromium:627816

Review-Url: https://codereview.webrtc.org/2606033002
Cr-Commit-Position: refs/heads/master@{#16188}
2017-01-20 14:14:25 +00:00
2d4d653e1f Fix msan flake in rtcstats_integrationtest.cc.
This CL https://codereview.webrtc.org/2641763003 changed echo return
loss /...enhancement stats from being optional to being undefined
because that was the observed behavior (and a TODO was added to
investigate why).

It turns out that these stats are sometimes available, e.g. if the test
runs for a while like MSAN bot does, so this turned the test flaky.
Example failure:
https://build.chromium.org/p/client.webrtc/builders/Linux%20MSan/builds/8242

This CL reverts that change without reverting the rest of the CL which
other CLs depend on, and updates the TODO.

BUG=chromium:627816
TBR=hta@webrtc.org
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2640743007
Cr-Commit-Position: refs/heads/master@{#16187}
2017-01-20 12:16:41 +00:00
c854ac3755 Stop camera onStop instead of onPause.
In multi-window mode the non-active activity receives onPause. We
shouldn't stop the camera in this case.

BUG=webrtc:7018

Review-Url: https://codereview.webrtc.org/2648483003
Cr-Commit-Position: refs/heads/master@{#16186}
2017-01-20 12:09:11 +00:00
42f6d2fb6c RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector.
VideoReceiverInfo::frames_received added based on
VideoReceiveStream::Stats::frame_counts (.key_frames + .delta_frames).

BUG=webrtc:6757, chromium:659137, chromium:627816

Review-Url: https://codereview.webrtc.org/2607913002
Cr-Commit-Position: refs/heads/master@{#16185}
2017-01-20 11:56:50 +00:00
7319f26632 Roll chromium_revision 780d18a4ff..7649e76842 (445004:445027)
Change log: 780d18a4ff..7649e76842
Full diff: 780d18a4ff..7649e76842

Changed dependencies:
* src/build: 00810858e7..43ba3a29da
* src/third_party: 903ea20111..e6dcf7494f
* src/tools: a23f646628..5012bd139b
DEPS diff: 780d18a4ff..7649e76842/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2648743003
Cr-Commit-Position: refs/heads/master@{#16184}
2017-01-20 11:28:46 +00:00
30fe5e0d10 Prevent downstream linter warnings.
BUG=None

Review-Url: https://codereview.webrtc.org/2643853007
Cr-Commit-Position: refs/heads/master@{#16183}
2017-01-20 11:10:08 +00:00
35564065ca Camera1Session: Fix camera sometimes getting stopped twice.
Moves setting state as stopped to stopInternal. Checks that state is not
stopped in stopInternal.

BUG=webrtc:7015

Review-Url: https://codereview.webrtc.org/2640093003
Cr-Commit-Position: refs/heads/master@{#16182}
2017-01-20 11:09:03 +00:00
9e30274c03 RTCMediaStreamTrackStats collected on a per-attachment basis.
According to recent spec change:
https://github.com/w3c/webrtc-stats/pull/138/files

This establishes the relationship between tracks and
[Voice/Video][Sender/Receiver]Info(s). Follow-up CLs will easily be able
to collect more stats from them.

BUG=webrtc:6757, chromium:659137, chromium:627816

Review-Url: https://codereview.webrtc.org/2641763003
Cr-Commit-Position: refs/heads/master@{#16181}
2017-01-20 10:47:10 +00:00
fd6c94d311 Allow more config changes for CallActivity.
CallActivity doesn't survive recreation so we have to handle these config
changes. Fortunately, they don't seem to require any special handling.

BUG=webrtc:7018

Review-Url: https://codereview.webrtc.org/2645853002
Cr-Commit-Position: refs/heads/master@{#16180}
2017-01-20 10:36:49 +00:00
3e9229045e Load library dependencies in AppRTCMobile.
Older Android versions have trouble locating the library dependencies
automatically. Loading them manually resolves the issue.

BUG=webrtc:6751

Review-Url: https://codereview.webrtc.org/2635233002
Cr-Commit-Position: refs/heads/master@{#16179}
2017-01-20 09:46:50 +00:00
be850e1b1d Clear out cached codecs when calculating new codec lists.
Without this, every time WebRtcVideoEngine2 calls supported_codecs(),
the codec list grows.

BUG=webrtc:7020

Review-Url: https://codereview.webrtc.org/2639423006
Cr-Commit-Position: refs/heads/master@{#16178}
2017-01-20 09:07:26 +00:00
204030a256 Roll chromium_revision bdeae63b37..780d18a4ff (444971:445004)
Change log: bdeae63b37..780d18a4ff
Full diff: bdeae63b37..780d18a4ff

Changed dependencies:
* src/base: f6f8cd35e1..4f9b6b4f4e
* src/testing: 2787e29b16..c11b06231a
* src/third_party: 8a2ee6884d..903ea20111
* src/tools: b6ebe7691d..a23f646628
DEPS diff: bdeae63b37..780d18a4ff/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2642263003
Cr-Commit-Position: refs/heads/master@{#16177}
2017-01-20 07:59:52 +00:00
a01d2f5cf7 Roll chromium_revision 34215edf2e..bdeae63b37 (444898:444971)
Change log: 34215edf2e..bdeae63b37
Full diff: 34215edf2e..bdeae63b37

Changed dependencies:
* src/base: e7d871b6b2..f6f8cd35e1
* src/testing: cd285cd30d..2787e29b16
* src/third_party: 56f93b1301..8a2ee6884d
* src/third_party/catapult: fe2fa4bc83..49e3f62b24
* src/third_party/libFuzzer/src: e6cbbd6ba1..78ee52d0c6
* src/tools: 071d8aeee2..b6ebe7691d
DEPS diff: 34215edf2e..bdeae63b37/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2644673004
Cr-Commit-Position: refs/heads/master@{#16176}
2017-01-20 05:00:21 +00:00
888874f761 Allow GCC 4.9 to compile Chromium
In order to implicit cast an lvalue to an rvalue when returning
from a function, the return type and type of variable in the return
statement previously had to be exactly the same. When this was not
the case, std::move was required. For instance, when returning a
std::unique_ptr<Derived> variable in a function with a
std::unique_ptr<Base> return type, std::move is required.

DR 1579 changed this, and allows for implicitly converting
to the return type, if the return type has a constructor(T&&), where
T is the type of the local variable being returned. DR 1579 was
implemented in GCC 5, but not in GCC 4.9 and below. By explicitly
qualifying the local variable with std::move, we allow for compiling
with GCC 4.9 and incur no performance penalty. The code is still
absolutely correct to the word of C++11.

BUG=chromium:682965

See also:
* https://bugs.gentoo.org/show_bug.cgi?id=600288
* https://stackoverflow.com/questions/22018115/converting-stdunique-ptrderived-to-stdunique-ptrbase#comment33375875_22018521
* http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2014/n3833.html#1579

Review-Url: https://codereview.webrtc.org/2642053003
Cr-Commit-Position: refs/heads/master@{#16175}
2017-01-20 04:20:45 +00:00
8944ab3fec Roll chromium_revision 1a7fcf6220..34215edf2e (444851:444898)
Change log: 1a7fcf6220..34215edf2e
Full diff: 1a7fcf6220..34215edf2e

Changed dependencies:
* src/third_party: 29884dce7f..56f93b1301
* src/tools: b8c8920f3b..071d8aeee2
DEPS diff: 1a7fcf6220..34215edf2e/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2640323005
Cr-Commit-Position: refs/heads/master@{#16174}
2017-01-20 01:44:09 +00:00
b2cdd93fd6 Remove the dependency of TransportChannel and TransportChannelImpl.
DtlsTransportChannelWrapper is renamed to be DtlsTransport which inherits from
DtlsTransportInternal. There will be no concept of "channel" in p2p level.
Both P2PTransportChannel and DtlsTransport don't depend on TransportChannel
and TransportChannelImpl any more and they are removed in this CL.

BUG=none

Review-Url: https://codereview.webrtc.org/2606123002
Cr-Commit-Position: refs/heads/master@{#16173}
2017-01-20 00:54:25 +00:00
9d643e87af Roll chromium_revision 113278e435..1a7fcf6220 (444801:444851)
Change log: 113278e435..1a7fcf6220
Full diff: 113278e435..1a7fcf6220

Changed dependencies:
* src/build: 4c65f6261e..00810858e7
* src/testing: 490308d809..cd285cd30d
* src/third_party: 34af144390..29884dce7f
* src/third_party/catapult: 1e05d2f840..fe2fa4bc83
* src/tools: d5a537f227..b8c8920f3b
DEPS diff: 113278e435..1a7fcf6220/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2644793005
Cr-Commit-Position: refs/heads/master@{#16172}
2017-01-19 22:36:54 +00:00
537798b7e9 Roll chromium_revision d50ce8a895..113278e435 (444743:444801)
Change log: d50ce8a895..113278e435
Full diff: d50ce8a895..113278e435

Changed dependencies:
* src/build: 70270c163a..4c65f6261e
* src/third_party: 32de94da6c..34af144390
* src/tools: 7e4f4b7490..d5a537f227
DEPS diff: d50ce8a895..113278e435/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2643093002
Cr-Commit-Position: refs/heads/master@{#16171}
2017-01-19 20:08:34 +00:00
9410b51037 GN: Add audio_conference_mixer_unittests to modules_unittests.
Was removed by accident in https://codereview.webrtc.org/2629923002/

BUG=webrtc:6954
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2641203002
Cr-Commit-Position: refs/heads/master@{#16170}
2017-01-19 19:57:56 +00:00
d74886350e Fix PseudoTcp to handle incoming packets with invalid SEQ field
Previously PseudoTcp::process() didn't handle the case when the peer
sends a packet that's outside of the receive window, which was causing
DCHECK failures in the fuzzer.

BUG=681849

Review-Url: https://codereview.webrtc.org/2640173002
Cr-Commit-Position: refs/heads/master@{#16169}
2017-01-19 18:53:35 +00:00