233 Commits

Author SHA1 Message Date
agrieve
babf8ee78c Add test runner scripts for instrumentation tests
BUG=599919
NOTRY=True
TBR=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1863363002

Cr-Commit-Position: refs/heads/master@{#12294}
2016-04-08 13:54:32 +00:00
nisse
2ded9b19d1 Replace SetCapturer and SetCaptureDevice by SetSource.
Drop return value.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1766653002

Cr-Commit-Position: refs/heads/master@{#12291}
2016-04-08 09:24:01 +00:00
skvlad
e0d4637bea Allow applications to control audio send bitrate through RtpParameters.
This change builds on top of the refactoring in https://codereview.webrtc.org/1841083008/, and enables WebRTC client applications to control the max send bitrate for every audio stream through RtpParameters.

The AudioSendStream now stores the last codec spec, and whenever a global or per-stream bitrate limit changes, the effective limit (smaller of the two) is recomputed and the codec is reconfigured with that bitrate.

TBR=pthatcher
BUG=

Review URL: https://codereview.webrtc.org/1847353004

Cr-Commit-Position: refs/heads/master@{#12290}
2016-04-08 05:59:32 +00:00
Taylor Brandstetter
05697a62fc Gracefully continue if ConnectivityManager.requestNetwork fails.
This happens on Android 6.0.0, which requires the WRITE_SETTINGS
permission, which is somewhat impractical to obtain.

R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1863413003 .

Cr-Commit-Position: refs/heads/master@{#12288}
2016-04-08 00:36:55 +00:00
Minyue
2a8a78c905 Add AEC filter divergence metric to StatsCollector.
A new metric that tells how often the AEC linear filter diverges has been recently introduced, see
https://codereview.webrtc.org/1739993003/

This metric can reflect echo failure and ducking.

In this CL, we add a field in StatsCollector to receive this metric.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1866983002 .

Cr-Commit-Position: refs/heads/master@{#12282}
2016-04-07 14:47:53 +00:00
nisse
5b68ab50bb Extended proxy abstraction, to call certain methods to the worker thread.
Extracted from cl https://codereview.webrtc.org/1766653002/, where
AddOrUpdateSink results in a deadlock.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1861633002

Cr-Commit-Position: refs/heads/master@{#12281}
2016-04-07 14:46:00 +00:00
Magnus Jedvert
fae14804ee Android: Fix chroma sampling bug when converting from texture to YUV
The xUnit for the UV channels in SurfaceTextureHelper.YuvConverter is
currently calculated from 1 / (2 * width). It should be 1 / (width / 2)
instead.

R=nisse@webrtc.org

Review URL: https://codereview.webrtc.org/1862003002 .

Cr-Commit-Position: refs/heads/master@{#12274}
2016-04-07 07:26:54 +00:00
kwiberg
b4d01c4ded A bunch of interfaces: Return scoped_ptr<SSLCertificate>
Instead of using a raw pointer output parameter. This affects

  SSLStreamAdapter::GetPeerCertificate
  Transport::GetRemoteSSLCertificate
  TransportChannel::GetRemoteSSLCertificate
  TransportController::GetRemoteSSLCertificate
  WebRtcSession::GetRemoteSSLCertificate

This is a good idea in general, but will also be very convenient when
scoped_ptr is gone, since unique_ptr doesn't have an .accept() method.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1802013002

Cr-Commit-Position: refs/heads/master@{#12262}
2016-04-06 12:15:11 +00:00
magjed
38d653c927 Revert of Switch to using EGL 1.0 for rendering and HW codec. (patchset #1 id:1 of https://codereview.webrtc.org/1829923002/ )
Reason for revert:
EGL 1.4 was not the cause of the deadlock. See https://bugs.chromium.org/p/webrtc/issues/detail?id=5702 for more info.

Original issue's description:
> Switch to using EGL 1.0 for rendering and HW codec.
>
> Using EGL 1.4 may cause texture rendering deadlock on some
> Android devices.
>
> R=jiayl@webrtc.org
>
> Committed: 887a19b9d2

BUG=webrtc:5702
TBR=jiayl@webrtc.org,glaznev@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.

Review URL: https://codereview.webrtc.org/1866653002

Cr-Commit-Position: refs/heads/master@{#12257}
2016-04-06 09:26:30 +00:00
perkj
05255b0e8a Revert of Changed P2PTestConductor to use a separate WorkerThread. (patchset #1 id:1 of https://codereview.webrtc.org/1859933002/ )
Reason for revert:
Causes P2PTestConductor.LocalP2PTestDtlsTransferCaller to fail on Win dbg.

https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug/builds/7469/steps/peerconnection_unittests/logs/stdio

e:\b\build\slave\win\build\src\webrtc\api\peerconnection_unittest.cc(1221): error: Value of: initiating_client_->ice_connection_state()
  Actual: 2
Expected: webrtc::PeerConnectionInterface::kIceConnectionCompleted
Which is: 3

Original issue's description:
> Changed P2PTestConductor to use a separate WorkerThread.
>
> P2PTestConductor currently use the current thread both as a signaling thread and a worker thread. Although convenient while debugging, it can also hide real bugs. An example is https://codereview.webrtc.org/1766653002/#ps420001 where the worker thread is deadlocked in the track proxy due to that the worker thread waits for the signaling thread but the proxy in turns invokes the worker thread..... That bug was only discovered on Android. I suggest we let the P2PTestConductor use a separate thread as a worker thread to better cover how PeerConnections are used in reality.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/6172401972c54813698d73580779d675d99178b4
> Cr-Commit-Position: refs/heads/master@{#12252}

TBR=nisse@webrtc.org,pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1866503003

Cr-Commit-Position: refs/heads/master@{#12255}
2016-04-06 08:28:34 +00:00
perkj
efc38584b7 Remove deprecated MediaStreamTrackInterface::set_state
Chrome is cleaned up in https://codereview.chromium.org/1853793002/
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1854063002

Cr-Commit-Position: refs/heads/master@{#12254}
2016-04-06 08:16:00 +00:00
perkj
6172401972 Changed P2PTestConductor to use a separate WorkerThread.
P2PTestConductor currently use the current thread both as a signaling thread and a worker thread. Although convenient while debugging, it can also hide real bugs. An example is https://codereview.webrtc.org/1766653002/#ps420001 where the worker thread is deadlocked in the track proxy due to that the worker thread waits for the signaling thread but the proxy in turns invokes the worker thread..... That bug was only discovered on Android. I suggest we let the P2PTestConductor use a separate thread as a worker thread to better cover how PeerConnections are used in reality.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1859933002

Cr-Commit-Position: refs/heads/master@{#12252}
2016-04-06 07:03:07 +00:00
Per
766ad3b989 This cl do a major cleanup of the VideoAdapter and make sure it does care about the VideoSinkWants.max_pixel_count and VideoSinkWants.max_pixel_count_step_up.
Unit tests are updated to test that screen share is not adapted but it does not change the VideoSinkWants in WebRtcVideoEngine2::SendStream due to a switch to screen share. The reason is that it works anyway and sprang is looking into how to do adaptation based on frame rate as well and use the adapter for screen share as well.

BUG=webrtc:5688, webrtc:5426
R=nisse@webrtc.org, pbos@webrtc.org, sprang@google.com

Review URL: https://codereview.webrtc.org/1836043004 .

Cr-Commit-Position: refs/heads/master@{#12240}
2016-04-05 13:23:58 +00:00
magjed
9fdb6cf255 Andoid EglBase: Detect failure to find EGL config
BUG=b/27950559

Review URL: https://codereview.webrtc.org/1855953002

Cr-Commit-Position: refs/heads/master@{#12239}
2016-04-05 13:08:13 +00:00
magjed
23b08eb531 Android VideoCapture: Add null checks in stopCaptureOnCameraThread
If stopCapture is called shortly after startCapture, and the first startCaptureOnCameraThread failed, but still hasn't retried 3 times, stopCaptureOnCameraThread will be called in a state where the camera is not initialized. This CL adds null checks in stopCaptureOnCameraThread to avoid crashes.

BUG=b/27939867

Review URL: https://codereview.webrtc.org/1854103002

Cr-Commit-Position: refs/heads/master@{#12234}
2016-04-05 08:37:08 +00:00
tkchin
7d06a8cfe4 Add CoreVideoFrameBuffer.
- Makes vt h264 decoder output CoreVideoFrameBuffer
- Makes iOS renderer convert frame buffer if it is not i420

BUG=

Review URL: https://codereview.webrtc.org/1853503003

Cr-Commit-Position: refs/heads/master@{#12224}
2016-04-04 21:10:47 +00:00
deadbeef
60631775fa Allowing a Java object field to be null in a new JNI helper method.
Java objects in the API should be allowed to be null in some cases.
Specifically, a null value for maxBitrateBps in RtpParameters.java
has a specific meaning and doesn't imply an error has occurred.

NOTRY=True

Review URL: https://codereview.webrtc.org/1853523002

Cr-Commit-Position: refs/heads/master@{#12221}
2016-04-04 17:27:31 +00:00
Peter Boström
85829fd90c Make QualityScaler more responsive to downgrades.
Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
windows for going up quickly in the beginning of a call (before any
downscaling happens due to bad quality).

BUG=webrtc:5678
R=glaznev@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1830593003 .

Cr-Commit-Position: refs/heads/master@{#12219}
2016-04-04 16:11:18 +00:00
nisse
71a0c2f9a6 Deprecate GetWidth() and GetHeight() methods. Replaced by width() and height().
Delete GetChromaWidth, GetChromaHeight, and GetChromaSize.

Delete unused function VideoFrameEqual.

BUG=webrtc:5682

Review URL: https://codereview.webrtc.org/1838353004

Cr-Commit-Position: refs/heads/master@{#12213}
2016-04-04 07:57:37 +00:00
Taylor Brandstetter
7ff1737e7c Re-enabling tests that were disabled for Windows debug builds.
The issue should be fixed by this commit:
https://boringssl.googlesource.com/boringssl.git/+/feaa57d13daa0b5bf3c068ce18d24870d50bfae9

BUG=webrtc:5659
NOTRY=True
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1837393002 .

Cr-Commit-Position: refs/heads/master@{#12200}
2016-04-01 18:50:47 +00:00
nisse
fcc640f8f6 Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector,
without involving the VideoMediaChannel.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1827023002

Cr-Commit-Position: refs/heads/master@{#12193}
2016-04-01 08:10:50 +00:00
tkchin
89717aad50 Improve iOS frame capture threading.
- Posts to WebRTC thread instead of Send
- Sample buffers are returned on capture session queue instead of main queue
- Camera switch happens on captures session queue

BUG=webrtc:5679, webrtc:4212

Review URL: https://codereview.webrtc.org/1838933004

Cr-Commit-Position: refs/heads/master@{#12186}
2016-04-01 00:14:09 +00:00
tkchin
8b9ca953a4 Minor ObjC header updates.
BUG=

Review URL: https://codereview.webrtc.org/1845133002

Cr-Commit-Position: refs/heads/master@{#12183}
2016-03-31 19:08:12 +00:00
Per
c0d31e915c Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool>
It turns out that it is used as if it has three states: on/off default.
This reverts back to the behaviour prior to https://codereview.webrtc.org/1773993002

BUG=chromium:594434
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1842073002 .

Cr-Commit-Position: refs/heads/master@{#12181}
2016-03-31 15:23:53 +00:00
kjellander
00984ff688 Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
The re-land moves the isolate build targets for media.gyp
and pc.gyp into the include_tests==1 condition.
This has been tested in a Chromium checkout and no longer
causes the error that was seen after landing
https://codereview.webrtc.org/1839763004/

Original issue's description:
> Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ )
>
> Reason for revert:
> Breaks Chromium: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/11313/steps/gclient%20runhooks/logs/stdio:
>
> Updating projects from gyp files...
> Using overrides found in /Users/chrome-bot/.gyp/include.gypi
> Traceback (most recent call last):
>   File "src/build/gyp_chromium", line 12, in <module>
>     execfile(__file__ + '.py')
>   File "src/build/gyp_chromium.py", line 341, in <module>
>     sys.exit(main())
>   File "src/build/gyp_chromium.py", line 328, in main
>     gyp_rc = gyp.main(args)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 538, in main
>     return gyp_main(args)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 514, in gyp_main
>     options.duplicate_basename_check)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 130, in Load
>     params['parallel'], params['root_targets'])
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 2800, in Load
>     RemoveLinkDependenciesFromNoneTargets(targets)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 1510, in RemoveLinkDependenciesFromNoneTargets
>     if targets[t].get('variables', {}).get('link_dependency', 0):
> KeyError: '/b/build/slave/Mac_Builder/build/src/third_party/webrtc/media/media.gyp:rtc_media_unittests#target'
> Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/Mac_Builder/build
> Hook '/usr/bin/python src/build/gyp_chromium' took 20.29 secs
>
> Original issue's description:
> > Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
> >
> > These contributes to circular dependency problems in WebRTC
> > since one have to depend on webrtc.gyp in order to depend on
> > a target in them.
> >
> > This reduces the number of cyclic dependencies in WebRTC from 21
> > to 16.
> >
> > BUG=webrtc:4243
> > NOTRY=True
> > NOPRESUBMIT=True
> >
> > Committed: https://crrev.com/231b69f28dd22f4e2d98e5048f8eaae7b20915e6
> > Cr-Commit-Position: refs/heads/master@{#12166}
>
> TBR=pthatcher@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4243
>
> Committed: https://crrev.com/72644d2cf6b14bbc4a107f79158eaa225f3196b5
> Cr-Commit-Position: refs/heads/master@{#12167}

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243

Review URL: https://codereview.webrtc.org/1843193002

Cr-Commit-Position: refs/heads/master@{#12180}
2016-03-31 14:23:52 +00:00
Magnus Jedvert
3db6f9b4df Android EGL: Synchronize calls to eglSwapBuffers and eglMakeCurrent
BUG=webrtc:5702
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1848483002 .

Cr-Commit-Position: refs/heads/master@{#12178}
2016-03-31 11:17:20 +00:00
Niels Möller
71bdda0ede Add RTCConfiguration getter and setter methods. The immediate plan is to move some flags into an embedded MediaConfig (https://codereview.webrtc.org/1818033002/), which will be possible after Chrome is updated to use these new setter methods.
BUG=webrtc:4906
R=hbos@google.com, hbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1836083003 .

Cr-Commit-Position: refs/heads/master@{#12177}
2016-03-31 11:00:08 +00:00
magjed
82b750b80e Android SurfaceTextureHelper: Distinguish thread names for decoder and camera
Review URL: https://codereview.webrtc.org/1843973002

Cr-Commit-Position: refs/heads/master@{#12176}
2016-03-31 07:54:18 +00:00
kjellander
72644d2cf6 Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ )
Reason for revert:
Breaks Chromium: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/11313/steps/gclient%20runhooks/logs/stdio:

Updating projects from gyp files...
Using overrides found in /Users/chrome-bot/.gyp/include.gypi
Traceback (most recent call last):
  File "src/build/gyp_chromium", line 12, in <module>
    execfile(__file__ + '.py')
  File "src/build/gyp_chromium.py", line 341, in <module>
    sys.exit(main())
  File "src/build/gyp_chromium.py", line 328, in main
    gyp_rc = gyp.main(args)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 538, in main
    return gyp_main(args)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 514, in gyp_main
    options.duplicate_basename_check)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 130, in Load
    params['parallel'], params['root_targets'])
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 2800, in Load
    RemoveLinkDependenciesFromNoneTargets(targets)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 1510, in RemoveLinkDependenciesFromNoneTargets
    if targets[t].get('variables', {}).get('link_dependency', 0):
KeyError: '/b/build/slave/Mac_Builder/build/src/third_party/webrtc/media/media.gyp:rtc_media_unittests#target'
Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/Mac_Builder/build
Hook '/usr/bin/python src/build/gyp_chromium' took 20.29 secs

Original issue's description:
> Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
>
> These contributes to circular dependency problems in WebRTC
> since one have to depend on webrtc.gyp in order to depend on
> a target in them.
>
> This reduces the number of cyclic dependencies in WebRTC from 21
> to 16.
>
> BUG=webrtc:4243
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/231b69f28dd22f4e2d98e5048f8eaae7b20915e6
> Cr-Commit-Position: refs/heads/master@{#12166}

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243

Review URL: https://codereview.webrtc.org/1846693002

Cr-Commit-Position: refs/heads/master@{#12167}
2016-03-30 18:54:14 +00:00
kjellander
231b69f28d Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
These contributes to circular dependency problems in WebRTC
since one have to depend on webrtc.gyp in order to depend on
a target in them.

This reduces the number of cyclic dependencies in WebRTC from 21
to 16.

BUG=webrtc:4243
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1839763004

Cr-Commit-Position: refs/heads/master@{#12166}
2016-03-30 18:40:38 +00:00
jbauch
f1f87203d7 Split ByteBuffer into writer/reader objects.
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.

BUG=webrtc:5155,webrtc:5670

Review URL: https://codereview.webrtc.org/1821083002

Cr-Commit-Position: refs/heads/master@{#12160}
2016-03-30 13:43:44 +00:00
Taylor Brandstetter
f8711c0209 Adding JNI binding for 'active' field in RTP encodings.
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1834223004 .

Cr-Commit-Position: refs/heads/master@{#12153}
2016-03-30 00:21:40 +00:00
Taylor Brandstetter
d45b95c270 Making new unit test assertions use the standard timeout.
This is to reduce the likelihood of them flaking in the future.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1836253002 .

Cr-Commit-Position: refs/heads/master@{#12149}
2016-03-29 20:16:55 +00:00
kwiberg
9708e9c599 Don't call operator== with scoped_ptr<T> and T*
That won't work when rtc::scoped_ptr becomes a type alias for
std::unique_ptr.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1834103002

Cr-Commit-Position: refs/heads/master@{#12145}
2016-03-29 17:17:45 +00:00
magjed
2378212e5a Android HW decoder: Add support for textures when using EGL 1.0
There is no reason to require EGL 1.4 for the decoder, it works fine with EGL 1.0 as well. It's important to support EGL 1.0 after https://codereview.webrtc.org/1829923002

BUG=webrtc:5702
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1837913005

Cr-Commit-Position: refs/heads/master@{#12136}
2016-03-29 12:13:29 +00:00
Peter Boström
ae69b02846 Fix typo in FakeAdmTest.TestProcess name.
BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1820133003 .

Cr-Commit-Position: refs/heads/master@{#12134}
2016-03-29 10:13:07 +00:00
Per
85e46a8f13 Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates
Increase delay for waiting for AudioSource state change to 10ms to hopefully fix flake that started happening on the 24th of Mars after landing https://codereview.webrtc.org/1816143002/.

TBR=deadbeef@webrtc.org
BUG=5692

Review URL: https://codereview.webrtc.org/1844503002 .

Cr-Commit-Position: refs/heads/master@{#12130}
2016-03-29 07:18:16 +00:00
skvlad
303b3c21a4 Added the JNI interface to get and set RtpParameters and the maximum bitrate limits.
Defined a JavaCollection convenience class to simplify iterating over collections from within JNI code
Follow-up to https://codereview.webrtc.org/1788583004/.
BUG=

Review URL: https://codereview.webrtc.org/1819553002

Cr-Commit-Position: refs/heads/master@{#12125}
2016-03-25 02:36:54 +00:00
perkj
7ca142e047 ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy.
https://codereview.webrtc.org/1816143002/ broke the Chrome test builds. The method is only used for tests and adding a dummy implementation seem to be enough.

BUG=webrtc:5426
R=guidou@webrtc.org

Review URL: https://codereview.webrtc.org/1828233002 .

Cr-Commit-Position: refs/heads/master@{#12118}
2016-03-24 13:19:24 +00:00
perkj
4a3a135e68 More cleanup of cricket::VideoCapturer
Remove unused state CS_PAUSED and dummy implementation of Pause()
Remove unused SignalVideoFrame

This should now be possible since https://codereview.chromium.org/1745923002/ has landed.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1832793002

Cr-Commit-Position: refs/heads/master@{#12116}
2016-03-24 11:39:56 +00:00
perkj
d61bf803d2 Removed MediaStreamTrackInterface::set_state
The track state should be implicitly set by the underlying source.
This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed.

Note that this cl depend on https://codereview.webrtc.org/1790633002.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1816143002

Cr-Commit-Position: refs/heads/master@{#12115}
2016-03-24 10:16:23 +00:00
honghaiz
a0c44eaa82 Add 16-bit network id to the candidate signaling.
Also include that in the stun-ping request as part of the
network-info attribute.
Change the network cost to be 16 bits.

BUG=

Review URL: https://codereview.webrtc.org/1815473002

Cr-Commit-Position: refs/heads/master@{#12110}
2016-03-23 23:07:54 +00:00
Alex Glaznev
887a19b9d2 Switch to using EGL 1.0 for rendering and HW codec.
Using EGL 1.4 may cause texture rendering deadlock on some
Android devices.

R=jiayl@webrtc.org

Review URL: https://codereview.webrtc.org/1829923002 .

Cr-Commit-Position: refs/heads/master@{#12109}
2016-03-23 22:01:54 +00:00
tkchin
24a62d5d83 Remove WEBRTC_IOS from RTCPeerConnectionFactory public header.
We shouldn't make external users define this flag to use our file.

BUG=

Review URL: https://codereview.webrtc.org/1825713003

Cr-Commit-Position: refs/heads/master@{#12106}
2016-03-23 18:29:32 +00:00
Taylor Brandstetter
a8415fe9ea Adding comments about threading around CreatePeerConnectionFactory.
This has confused a lot of developers (understandably).

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1828463002 .

Cr-Commit-Position: refs/heads/master@{#12105}
2016-03-23 17:38:16 +00:00
nisse
1509fa1aa9 Delete cricket::VideoRenderer.
TBR=glaznev@webrtc.org (deleting an #include in main_wnd.h)
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1819103003

Cr-Commit-Position: refs/heads/master@{#12101}
2016-03-23 11:06:05 +00:00
Niels Möller
8f59762897 Delete VideoRendererInterface.
Use in chromium was deleted a few days ago.

BUG=webrtc:5426
R=magjed@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1817473002 .

Cr-Commit-Position: refs/heads/master@{#12099}
2016-03-23 09:33:19 +00:00
perkj
c8f952deaa Propagate MediaStreamSource state to video tracks the same way as audio.
Also removes unused track states kLive and kFailed.
Since this also required a Video source to exist in all unit tests that create a track, a FakeVideoTrackSource is added and used in tests.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1790633002

Cr-Commit-Position: refs/heads/master@{#12098}
2016-03-23 07:34:01 +00:00
deadbeef
dbe2b8744f Adding support for RTCRtpEncodingParameters.active flag.
This will allow a sender to stop/start sending media on the
application's demand.

Among other things, this can allow an application to set a track on a
sender while the encoding(s) are inactive, allowing the encoder to be
initialized for that track, then later set the encodings to "active"
to instantly start sending the track.

Review URL: https://codereview.webrtc.org/1822923002

Cr-Commit-Position: refs/heads/master@{#12094}
2016-03-22 22:42:07 +00:00
Peter Boström
01bcbd0df6 Make Android min-resolution rotation-agnostic.
Min resolution shouldn't have anything to do with CVO being enabled or
not, nor device rotation.

BUG=webrtc:5678
R=glaznev@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1824083002 .

Cr-Commit-Position: refs/heads/master@{#12092}
2016-03-22 20:44:43 +00:00