7 Commits

Author SHA1 Message Date
perkj
f2880a0e04 Change webrtc::VideoSourceInterface to inherit rtc::VideoSourceInterface.
Also introduce a typedef VideoTrackSourceInterface to be able to start changing clients such as Chrome to use the name VideoTrackSourceInterface.

Document: https://docs.google.com/a/google.com/document/d/1mEIw_0uDzyHjL3l8a82WKp6AvgR8Tlwn9JGvhbUjVpY/edit?usp=sharing

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1758223002

Cr-Commit-Position: refs/heads/master@{#11854}
2016-03-03 09:51:56 +00:00
Per
0f13ec1265 Removed VideoSource dependency to ChannelManager.
Instead VideoSource directly access the cricket::VideoCapturer via the worker_thread.

Document: https://docs.google.com/a/google.com/document/d/1mEIw_0uDzyHjL3l8a82WKp6AvgR8Tlwn9JGvhbUjVpY/edit?usp=sharing

BUG=webrtc:5426
R=nisse@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1759473003 .

Cr-Commit-Position: refs/heads/master@{#11852}
2016-03-03 08:22:41 +00:00
Niels Möller
60653ba3cc New flag is_screencast in VideoOptions.
This cl copies the value of cricket::VideoCapturer::IsScreencast into
a flag in VideoOptions. It is passed on via the chain

VideortpSender::SetVideoSend
WebRtcVideoChannel2::SetVideoSend
WebRtcVideoChannel2::SetOptions
WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions

Where it's used, in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up
in parameters_, instead of calling capturer_->IsScreencast().

Doesn't touch screencast logic related to cpu adaptation, since that
code is in flux in a different cl.

Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options.

In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests.

Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters.

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

Review URL: https://codereview.webrtc.org/1711763003 .

Cr-Commit-Position: refs/heads/master@{#11837}
2016-03-02 10:41:49 +00:00
kjellander@webrtc.org
9b8df25c73 Move talk/session/media -> webrtc/pc
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
2016-02-12 05:48:10 +00:00
kjellander@webrtc.org
5ad129741c Rename webrtc/media/webrtc -> webrtc/media/engine
BUG=webrtc:5420
NOTRY=True
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1684163002 .

Cr-Commit-Position: refs/heads/master@{#11591}
2016-02-12 05:39:50 +00:00
kjellander
b24317bfda Fix license headers in webrtc/api.
In addition to the code moved from talk/app/webrtc
there were some files in webrtc/api/objctests that still
had the libjingle license header.

BUG=webrtc:5418
TBR=tkchin@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1680293005

Cr-Commit-Position: refs/heads/master@{#11552}
2016-02-10 15:54:53 +00:00
Henrik Kjellander
15583c19d7 Move talk/app/webrtc to webrtc/api
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
2016-02-10 09:53:26 +00:00