613 Commits

Author SHA1 Message Date
jbauch
ac8869ec5a Report metrics about negotiated ciphers.
This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).

BUG=428343

Review URL: https://codereview.webrtc.org/1156143005

Cr-Commit-Position: refs/heads/master@{#9537}
2015-07-03 08:36:22 +00:00
bemasc
0edd50ccb3 Support for onbufferedamountlow
Original review at https://webrtc-codereview.appspot.com/54679004/

BUG=https://code.google.com/p/chromium/issues/detail?id=496700

Review URL: https://codereview.webrtc.org/1207613006

Cr-Commit-Position: refs/heads/master@{#9527}
2015-07-01 20:34:42 +00:00
Zeke Chin
71f6f4405c iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.

BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1187573004.

Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 21:35:08 +00:00
Peter Thatcher
c0c3a865f4 Prevent JS from bypassing RTP data channel bandwidth limitation.
Normally the RTP data channel is capped at 30kbps, but by mangling the
SDP string, one could get around this limitation. With this fix,
SdpDeserialize will return an error if it detects this condition.

BUG=280726
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1196403004.

Cr-Commit-Position: refs/heads/master@{#9499}
2015-06-24 22:31:35 +00:00
magjed
59a677ada2 Android VideoRendererGui: Refactor GLES rendering
This CL should not change any visible behaviour. It does the following:
 * Extract GLES rendering into separate class GlRectDrawer. This class is also needed for future video encode with OES texture input.
 * Clean up current ScalingType -> display size calculation and introduce new SCALE_ASPECT_BALANCED (b/21735609) and remove unused SCALE_FILL.
 * Replace current mirror/rotation index juggling with android.opengl.Matrix operations instead.

Review URL: https://codereview.webrtc.org/1191243005

Cr-Commit-Position: refs/heads/master@{#9496}
2015-06-24 10:59:43 +00:00
phoglund
7ab5f801dd Adding an equals method for KeyValuePair for easier testing.
With this we can write stuff like

assertThat(result.mandatory,
    hasItem(new KeyValuePair("minWidth", "1280")));

The above will currently fail because the object falls back to ==.

BUG=None

Review URL: https://codereview.webrtc.org/1193883006

Cr-Commit-Position: refs/heads/master@{#9494}
2015-06-24 08:11:51 +00:00
Joachim Bauch
66f920ea57 Remove definition of non-existent method.
The private method "CreateDefaultLocalDescription" is defined in the
class, but not implemented or used anywhere.

R=juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1182793004.

Cr-Commit-Position: refs/heads/master@{#9493}
2015-06-24 07:34:41 +00:00
tommi
39b31001d2 Change kEchoCancellation to be 'echoCancellation'.
This is the second cl in WebRTC for this change and will be landed after Chromium has been updated to use kGooglEchoCancellation where that variant is required. See also the first part: https://codereview.webrtc.org/1179233003

BUG=webrtc:4747
R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1185963003

Cr-Commit-Position: refs/heads/master@{#9490}
2015-06-23 16:50:50 +00:00
jbauch
be24c94c95 Set / verify stats report timestamps.
This CL updates the track report timestamps which were fixed at "0" before
and updates the timestamps in reports for local audio tracks.

Also the timestamps are checked in various tests to make sure no "0" is
returned.

Original CL is at https://webrtc-codereview.appspot.com/51829004/

BUG=webrtc:4316
TBR=hta@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1204493002

Cr-Commit-Position: refs/heads/master@{#9485}
2015-06-22 22:06:50 +00:00
henrika
1d34fe979c Adds support for webrtc::test::ResourcePath on iOS
BUG=webrtc:4752
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1178843002.

Cr-Commit-Position: refs/heads/master@{#9445}
2015-06-16 08:04:24 +00:00
Henrik Lundin
b02af18c5c Follow-up: Remove old DelayCorrection AEC config
This is a follow-up to r9401, where the configuration DelayCorrection
was replaced by ExtendedFilter.

This change also removes the media constraint
kExperimentalEchoCancellation which was replaced by
kExtendedFilterEchoCancellation in the same CL.

Both settings that are now being removed were kept in the code to avoid
API breakages. In https://codereview.chromium.org/1167343004,
depending code has been updated to avoid breakages.

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1181413004.

Cr-Commit-Position: refs/heads/master@{#9444}
2015-06-16 07:53:32 +00:00
Åsa Persson
2b679250fb VideoCapturerAndroid: Add possibility to request a new resolution from the video adapter.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1178643006.

Cr-Commit-Position: refs/heads/master@{#9434}
2015-06-15 07:53:16 +00:00
Tommi
70c7fe14ac Add kGoogEchoCancellation to MediaConstraintsInterface.
This constraint will be equal to kEchoCancellation until we've updated Chromium to use kGoogEchoCancellation where that constraint is needed.  Once that's done, I'll change kEchoCancellation to be 'echoCancellation'.

BUG=webrtc:4747
R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1179233003.

Cr-Commit-Position: refs/heads/master@{#9433}
2015-06-15 07:14:15 +00:00
Alex Glaznev
782671f798 Improve Android HW decoder error handling.
- Remove an option to use MediaCodec SW decoder from Java layer.
- Better handling Java exceptions in JNI - detect exceptions
and either try to reset the codec or fallback to SW decoder.
- If any error is reported by codec try to fallback to SW
codec for VP8 or reset decoder and continue decoding for H.264.
- Add more logging for error conditions.

R=wzh@webrtc.org

Review URL: https://codereview.webrtc.org/1178943007.

Cr-Commit-Position: refs/heads/master@{#9431}
2015-06-12 23:40:51 +00:00
Jon Hjelle
2f65ac1437 Fix crash and warning in AppRTCDemo
Don't dismiss the presented view controller if it's already being dismissed to clear a warning about dismissing from a view controller while a dismiss is in progress.

Remove the sample buffer delegate when capture is being stopped to avoid a crash when a delegate method is sent to a deallocated object.

BUG=webrtc:4734
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54669004.

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#9430}
2015-06-12 18:33:51 +00:00
Peter Kasting
728d9037c0 Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1172163004

Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 21:31:48 +00:00
Peter Kasting
b7e5054414 Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones.  For example:

* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps.  For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika

Review URL: https://codereview.webrtc.org/1168753002

Cr-Commit-Position: refs/heads/master@{#9419}
2015-06-11 19:56:03 +00:00
Magnus Jedvert
80cf97cddd Android rendering: Move common EGL and GL functions to separate classes
This CL does not make any functional changes. The purpose is to extract some common code that is needed for texture capture and texture encode.

This CL does the following changes:
* Move common EGL functions from org.webrtc.MediaCodecVideoDecoder to org.webrtc.EglBase.
* Move common GL functions from org.webrtc.VideoRendererGui to org.webrtc.GlUtil and org.webrtc.GlShader.
* Remove unused call to surfaceTexture.getTransformMatrix in YuvImageRenderer.
* Add helper functions rotatedWidth()/rotatedHeight() in VideoRenderer.I420Frame.

R=glaznev@webrtc.org, hbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47309005.

Cr-Commit-Position: refs/heads/master@{#9414}
2015-06-11 08:09:15 +00:00
Peter Kasting
f045e4da43 Prepare to convert various types to size_t.
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question.  This is preparation for a future change
that will convert a variety of types to size_t.

There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.

BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm

Review URL: https://codereview.webrtc.org/1174813003

Cr-Commit-Position: refs/heads/master@{#9413}
2015-06-11 04:15:51 +00:00
Tommi
532caeae2d Adding DCHECKs and constness to DtlsIdentityStore.
R=hbos@webrtc.org, hbos
BUG=

Review URL: https://codereview.webrtc.org/1171893003.

Cr-Commit-Position: refs/heads/master@{#9402}
2015-06-09 15:33:17 +00:00
Henrik Lundin
441f634731 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)

The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.

Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec."

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1151573021.

Cr-Commit-Position: refs/heads/master@{#9401}
2015-06-09 14:03:23 +00:00
Peter Boström
eb66e800d1 Re-land "Convert native handles to buffers before encoding."
This reverts commit a67675506c9057bd9ffd4d76aae8b743343d434d.

BUG=webrtc:4081
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1158273010

Cr-Commit-Position: refs/heads/master@{#9381}
2015-06-05 09:08:12 +00:00
Henrik Lundin
3fbf3f8841 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.

BUG=4696
TBR=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1166463006

Cr-Commit-Position: refs/heads/master@{#9380}
2015-06-05 09:04:20 +00:00
Henrik Lundin
5f4b7e2873 Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec.

BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54659004

Cr-Commit-Position: refs/heads/master@{#9378}
2015-06-05 07:55:40 +00:00
Tommi
a9952cdd0e Remove CHECK from GetThreadName.
It's safe for prctl() to fail, so we fall back on <noname> for thread names if we can't get one, instead of crashing.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57529004

Cr-Commit-Position: refs/heads/master@{#9363}
2015-06-03 16:59:24 +00:00
Wan-Teh Chang
6b990744d9 Revert "Import org.junit.Assert instead of junit.framework.Assert."
This reverts commit a88470964c55dc655022d1f46370565aa3be535f.

It broke Android builds:
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46: error: package org.junit does not exist
import static org.junit.Assert.*;
                       ^
TBR=glaznev@webrtc.org,pthatcher@webrtc.org
BUG=none

Review URL: https://webrtc-codereview.appspot.com/52039004

Cr-Commit-Position: refs/heads/master@{#9357}
2015-06-02 21:36:32 +00:00
Wan-Teh Chang
a88470964c Import org.junit.Assert instead of junit.framework.Assert.
This fixed the warning:
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46: warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;

R=glaznev@webrtc.org, pthatcher@webrtc.org
BUG=none

Review URL: https://webrtc-codereview.appspot.com/50209004

Cr-Commit-Position: refs/heads/master@{#9356}
2015-06-02 21:26:48 +00:00
Peter Boström
308d163c71 Revert "Convert native handles to buffers before encoding."
This reverts commit a831dc3a7d10a1fbaa258ee6b1ca6cfc7e91c5ca to unblock
rolling into Chromium.

BUG=4081
TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55549004

Cr-Commit-Position: refs/heads/master@{#9354}
2015-06-02 13:04:31 +00:00
Henrik Lundin
8e6fd46cc3 Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50179004

Cr-Commit-Position: refs/heads/master@{#9350}
2015-06-02 07:25:03 +00:00
Peter Boström
a831dc3a7d Convert native handles to buffers before encoding.
Required to permit conversion of NV12 handles on iOS to I420 for VP8
software encoding, which blocks texture-based capture. This change
enforces that all texture-based input provides a method for converting
native handles to I420 if they are ever used with software encoders that
do not understand the native handles.

BUG=4081
R=emircan@chromium.org, glaznev@webrtc.org, hbos@webrtc.org, magjed@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50909005

Cr-Commit-Position: refs/heads/master@{#9347}
2015-06-01 18:06:52 +00:00
Henrik Lundin
5263b3c1dd Add options for NetEq fast accelerate mode through libjingle
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.

When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.

BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55479004

Cr-Commit-Position: refs/heads/master@{#9344}
2015-06-01 08:29:55 +00:00
Miguel Casas-Sanchez
4765070b8d Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.

Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.

BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52629004

Cr-Commit-Position: refs/heads/master@{#9339}
2015-05-30 00:21:56 +00:00
Jon Hjelle
c2cb266c93 Match video orientation with device orientation for portrait and portrait upside down
BUG=
R=tkchin@webrtc.org

Committed: https://crrev.com/14c2695f2968d6e8546545a9b62940563073b4b6
Patch from Jon Hjelle <hjon@andynet.net>.

Cr-Commit-Position: refs/heads/master@{#9336}

Review URL: https://webrtc-codereview.appspot.com/55459004

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#9338}
2015-05-29 23:38:31 +00:00
Zeke Chin
7be99bdea1 Revert "Match video orientation with device orientation for portrait and portrait upside down"
Misspelt contributor's email address. Easier to revert and reland.
TBR=hjon@andyet.net

This reverts commit 14c2695f2968d6e8546545a9b62940563073b4b6.

BUG=

Review URL: https://webrtc-codereview.appspot.com/54619004

Cr-Commit-Position: refs/heads/master@{#9337}
2015-05-29 23:34:43 +00:00
Jon Hjelle
14c2695f29 Match video orientation with device orientation for portrait and portrait upside down
BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55459004

Patch from Jon Hjelle <hjon@andynet.net>.

Cr-Commit-Position: refs/heads/master@{#9336}
2015-05-29 22:25:00 +00:00
Zeke Chin
bc7dd7e023 Add RTCConfiguration constructor to RTCPeerConnection wrapper.
BUG=4658
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56419004

Cr-Commit-Position: refs/heads/master@{#9335}
2015-05-29 21:59:23 +00:00
Joachim Bauch
d935f912b1 Don't try to parse empty Ice urls.
https://crrev.com/7c4e7458b5ce99c13a75d5be7d718ef94e2f8f9f added support
to pass a list of urls for IceServer configurations. This CL fixes a
potential crash when empty urls are passed.

BUG=2096
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51969004

Cr-Commit-Position: refs/heads/master@{#9334}
2015-05-29 20:14:28 +00:00
Lally Singh
5c6c6e026b Implements TODOs for webrtc::datachannel state management when the SCTP association is congested. Adds missing state variables for each step in the transitions between DataChannelInterface::DataStates (kConnecting, kOpen, etc.), and uses them.
BUG=https://code.google.com/p/chromium/issues/detail?id=474650
R=jiayl@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44299004

Cr-Commit-Position: refs/heads/master@{#9331}
2015-05-29 15:52:44 +00:00
Jelena Marusic
c28a896a7b VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
BUG=4690

Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code

R=solenberg@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56499004

Cr-Commit-Position: refs/heads/master@{#9330}
2015-05-29 13:05:52 +00:00
Joachim Bauch
04e5b49827 Make maximum SSL version configurable through PeerConnectionFactory::Options
This can be used to activate DTLS 1.2 through a command-line flag from Chromium
later.

BUG=chromium:428343
R=jiayl@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/54509004

Cr-Commit-Position: refs/heads/master@{#9328}
2015-05-29 07:40:51 +00:00
Joachim Bauch
7c4e7458b5 Support multiple URLs in PeerConnectionInterface::IceServer
This adds support for multiple URLs in a IceServer configuration as
defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCIceServer.

BUG=2096
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57489004

Cr-Commit-Position: refs/heads/master@{#9320}
2015-05-28 21:06:44 +00:00
Donald Curtis
d4f769d8fc Stop video candidates getting down to audio.
Second attempt at adding a check to make sure that the video
transportproxy doesn't send down candidates to the audio transport
channel when things are bundled.

BUG=4665
R=juberti@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50059004

Cr-Commit-Position: refs/heads/master@{#9316}
2015-05-28 16:48:30 +00:00
Henrik Boström
3b187b9c0c Removed unnecessary includes of webrtcvideocapturer.h
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57469004

Cr-Commit-Position: refs/heads/master@{#9305}
2015-05-28 09:43:45 +00:00
Peter Boström
23c2e55479 Remove remaining .mk files.
These files are not supported, kept up to date or likely to build
anymore.

BUG=
R=glaznev@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46489004

Cr-Commit-Position: refs/heads/master@{#9303}
2015-05-28 09:05:11 +00:00
Tommi
0eefb4d5c3 Detach base/logging.* from base/stream.*.
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54529004

Cr-Commit-Position: refs/heads/master@{#9269}
2015-05-23 07:54:19 +00:00
Alejandro Luebs
4bf12eafba Revert "Fix sending wrong candidates down to transportchannel."
This reverts commit f65de8483e90d1d52d5d8f40f646e77bf45b10ea.

It was breaking the build bots: http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/3062

TBR=decurtis

BUG=

Review URL: https://webrtc-codereview.appspot.com/54539004

Cr-Commit-Position: refs/heads/master@{#9267}
2015-05-22 22:32:51 +00:00
Donald Curtis
f65de8483e Fix sending wrong candidates down to transportchannel.
BUG=4665
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54489004

Cr-Commit-Position: refs/heads/master@{#9266}
2015-05-22 21:55:26 +00:00
jackychen
98d8cf58ee Hardware VP8 encoding: Use QP as metric for resize.
Add vp8 frame header parser to get QP from vp8 bitstream.

BUG= 4273
R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49259004

Cr-Commit-Position: refs/heads/master@{#9256}
2015-05-21 18:11:53 +00:00
Peter Thatcher
af55ccc054 Add RtcpMuxPolicy support to PeerConnection.
BUG=4611
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/46169004

Cr-Commit-Position: refs/heads/master@{#9251}
2015-05-21 14:48:19 +00:00
Joachim Bauch
831c5585c7 Allow setting maximum protocol version for SSL stream adapters.
This CL adds an API to SSL stream adapters to set the maximum allowed
protocol version and with that implements support for DTLS 1.2.
With DTLS 1.2 the default cipher changes in the unittests as follows.

BoringSSL
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA -> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256

NSS
TLS_RSA_WITH_AES_128_CBC_SHA -> TLS_RSA_WITH_AES_128_GCM_SHA256

BUG=chromium:428343
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/50989004

Cr-Commit-Position: refs/heads/master@{#9232}
2015-05-20 10:48:24 +00:00