Many of these interfaces are not intuitive, or are the way they are for
complex historical reasons, so it would be nice to document these things
for future developers.
Also, many nonstandard things (such as RTCConfiguration options) were
not documented at all before this CL.
BUG=webrtc:7131
TBR=pthatcher@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2680273002
Cr-Commit-Position: refs/heads/master@{#16485}
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}
New features are:
- Invoke a destructor on the worker thread.
- Make proxy wrapper for a non-refcounted object.
- Ability to use unique_ptrs (as arguments or return values).
These features are needed by this CL:
https://codereview.webrtc.org/2632613002/
BUG=None
Review-Url: https://codereview.webrtc.org/2628343003
Cr-Commit-Position: refs/heads/master@{#16151}
Relanding again after fixing issue with RTC_DCHECKs.
This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13305}
Reason for revert:
Broke video sending for iOS AppRTCDemo. To repro, run iOS AppRTCDemo in Release in loopback mode. The revision prior to this change worked.
Original issue's description:
> Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: 2d5491783aTBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092273003
Cr-Commit-Position: refs/heads/master@{#13289}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13287}
Reason for revert:
Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland.
Original issue's description:
> Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c
> Cr-Commit-Position: refs/heads/master@{#13285}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2099843003
Cr-Commit-Position: refs/heads/master@{#13286}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13285}
The RtpReceiverObserverInterface is created.
The SignalFirstPacketReceived will be forwarded from BaseChannel to WebRtcSession.
WebRtcSession will forward SignalFirstAudioPacketReceived and SignalFirstVideoPacketReceived to the RtpReceiverInterface.
The application can listen to the Signal by implementing and registering a RtpReceiverObserver.
Review-Url: https://codereview.webrtc.org/1999853002
Cr-Commit-Position: refs/heads/master@{#13139}
This moves the implementation specific methods to separate classes
(RtpSenderInternal/RtpReceiverInternal) so that the interface classes
represent the interface that external applications can rely on.
The reason this wasn't done earlier was that PeerConnection needed
to store proxy pointers, but also needed to access implementation-
specific methods on the underlying objects. This is now possible
by using "RtpSenderProxyWithInternal<RtpSenderInternal>", which is a proxy
that implements RtpSenderInterface but also provides direct access
to an RtpSenderInternal.
Review-Url: https://codereview.webrtc.org/2023373002
Cr-Commit-Position: refs/heads/master@{#13056}
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1917193008 .
Cr-Commit-Position: refs/heads/master@{#12761}
And BEGIN_WORKER_PROXY_MAP --> BEGIN_PROXY_MAP.
This rename was suggested by Tommi, with the idea that a proxy
invoking methods on the worker thread should be the common case.
It's a followup to https://codereview.webrtc.org/1861633002/
This cl also adds unittests for proxy calls to the
worker thread.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1871833002
Cr-Commit-Position: refs/heads/master@{#12374}
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}
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}
