- The RTC_SUPPORTS_METAL macro allows consumers to gracefully handle compilation for different archs that are not supporting Metal.
BUG=webrtc:7079
Review-Url: https://codereview.webrtc.org/2722583002
Cr-Commit-Position: refs/heads/master@{#17004}
- The "flexfec-03" codec is advertised in the SDP whenever the
"WebRTC-FlexFEC-03-Advertised" field trial is enabled.
- Sending FlexFEC packets is enabled whenever the "flexfec-03" codec is
negotiated, and the "WebRTC-FlexFEC-03" field trial is enabled.
After this CL, the number of calls to
WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec during renegotiation
will be reduced for cases when only one endpoint has the "WebRTC-FlexFEC-03"
field trial enabled.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2716733005
Cr-Commit-Position: refs/heads/master@{#16925}
Reason for revert:
This CL breaks iOS AppRTCMobile. We don't have any automatic tests running on the bots yet, so please try AppRTCMobile locally before relanding.
Stack trace:
* thread #15: tid = 0x20e933, 0x0000000100488440 AppRTCMobile`webrtc::AudioRtpReceiver::OnFirstPacketReceived(this=0x0000000170156c60, channel=0x000000010511a600) + 48 at rtpreceiver.cc:133, name = 'Thread 0x0x10421b2a0', stop reason = EXC_BAD_ACCESS (code=1, address=0x1a1aac71979)
* frame #0: 0x0000000100488440 AppRTCMobile`webrtc::AudioRtpReceiver::OnFirstPacketReceived(this=0x0000000170156c60, channel=0x000000010511a600) + 48 at rtpreceiver.cc:133
frame #1: 0x000000010048a3f8 AppRTCMobile`void sigslot::_opaque_connection::emitter<webrtc::AudioRtpReceiver, cricket::BaseChannel*>(self=0x000000017424b380, args=0x000000010511a600) + 184 at sigslot.h:391
frame #2: 0x00000001005a30ec AppRTCMobile`void sigslot::_opaque_connection::emit<cricket::BaseChannel*>(this=0x000000017424b380, args=0x000000010511a600) const + 56 at sigslot.h:381
frame #3: 0x00000001005a3094 AppRTCMobile`sigslot::signal_with_thread_policy<sigslot::single_threaded, cricket::BaseChannel*>::emit(this=0x000000010511a678, args=0x000000010511a600) + 504 at sigslot.h:615
frame #4: 0x000000010057ef5c AppRTCMobile`sigslot::signal_with_thread_policy<sigslot::single_threaded, cricket::BaseChannel*>::operator(this=0x000000010511a678, args=0x000000010511a600)(cricket::BaseChannel*) + 32 at sigslot.h:621
frame #5: 0x000000010057ef00 AppRTCMobile`cricket::BaseChannel::OnMessage(this=0x000000010511a600, pmsg=0x000000016e676db0) + 600 at channel.cc:1494
frame #6: 0x0000000100584a58 AppRTCMobile`cricket::VoiceChannel::OnMessage(this=0x000000010511a600, pmsg=0x000000016e676db0) + 152 at channel.cc:1909
frame #7: 0x000000010017c0dc AppRTCMobile`rtc::MessageQueue::Dispatch(this=0x000000010421b2a0, pmsg=0x000000016e676db0) + 336 at messagequeue.cc:538
frame #8: 0x00000001001d8efc AppRTCMobile`rtc::Thread::ProcessMessages(this=0x000000010421b2a0, cmsLoop=-1) + 228 at thread.cc:496
frame #9: 0x00000001001d8e08 AppRTCMobile`rtc::Thread::Run(this=0x000000010421b2a0) + 28 at thread.cc:327
frame #10: 0x00000001001d8b0c AppRTCMobile`rtc::Thread::PreRun(pv=0x000000017000f030) + 300 at thread.cc:316
frame #11: 0x00000001843f1850 libsystem_pthread.dylib`_pthread_body + 240
frame #12: 0x00000001843f1760 libsystem_pthread.dylib`_pthread_start + 284
frame #13: 0x00000001843eed94 libsystem_pthread.dylib`thread_start + 4
Original issue's description:
> Add the OnAddTrack callback for Objective-C wrapper.
>
> Created an Obj-C wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
> The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
> The application can tell when tracks are added to the streams by listening to this callback.
>
> BUG=webrtc:6112
>
> Review-Url: https://codereview.webrtc.org/2513063003
> Cr-Commit-Position: refs/heads/master@{#16835}
> Committed: 633f6fe004TBR=tkchin@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2720753002
Cr-Commit-Position: refs/heads/master@{#16871}
Reason for revert:
Re-land
Original issue's description:
> Revert of Add metal view, shaders and renderer. (patchset #18 id:340001 of https://codereview.webrtc.org/2651743007/ )
>
> Reason for revert:
> Reverting due to breakage in the Google3 import
>
> Original issue's description:
> > Add metal view, shaders and renderer.
> >
> > This CL submits standalone Metal view, renderer and shader.
> >
> > BUG=webrtc:7079
> >
> > Review-Url: https://codereview.webrtc.org/2651743007
> > Cr-Commit-Position: refs/heads/master@{#16787}
> > Committed: fc8c97f950
>
> TBR=magjed@webrtc.org,kthelgason@webrtc.org,tkchin@webrtc.org,haysc@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7079
>
> Review-Url: https://codereview.webrtc.org/2711003004
> Cr-Commit-Position: refs/heads/master@{#16788}
> Committed: b681aabdfcTBR=magjed@webrtc.org,kthelgason@webrtc.org,tkchin@webrtc.org,haysc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7079
Review-Url: https://codereview.webrtc.org/2716703004
Cr-Commit-Position: refs/heads/master@{#16846}
Created an Obj-C wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2513063003
Cr-Commit-Position: refs/heads/master@{#16835}
This is a short term solution to change the volume of a RTCAudioTrack (which contains an RTCAudioSource property) until applyConstraints for RTCMediaStreamTracks has been implemented.
This CL adds one new Objective-C method to AudioSourceInterface's wrapper: -(void)setVolume:(double)volume
BUG=webrtc:6533, webrtc:6805
This is my first CL for Chromium/WebRTC, so please let me know if I did something wrong.
Review-Url: https://codereview.webrtc.org/2534843002
Cr-Commit-Position: refs/heads/master@{#16809}
Reason for revert:
Reverting due to breakage in the Google3 import
Original issue's description:
> Add metal view, shaders and renderer.
>
> This CL submits standalone Metal view, renderer and shader.
>
> BUG=webrtc:7079
>
> Review-Url: https://codereview.webrtc.org/2651743007
> Cr-Commit-Position: refs/heads/master@{#16787}
> Committed: fc8c97f950TBR=magjed@webrtc.org,kthelgason@webrtc.org,tkchin@webrtc.org,haysc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7079
Review-Url: https://codereview.webrtc.org/2711003004
Cr-Commit-Position: refs/heads/master@{#16788}
This CL submits standalone Metal view, renderer and shader.
BUG=webrtc:7079
Review-Url: https://codereview.webrtc.org/2651743007
Cr-Commit-Position: refs/heads/master@{#16787}
Adds a public init function in RTCVideoFrame that makes it possible to
create a frame from a CVPixelBufferRef.
BUG=webrtc:7177
NOTRY=True
Review-Url: https://codereview.webrtc.org/2700113003
Cr-Commit-Position: refs/heads/master@{#16746}
RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame, but it
currently contains some extra logic beyond that. We want RTCVideoFrame
to be as simple as possible, i.e. just a container with no extra state,
so we can use it as input to RTCVideoSource without complicating the
interface for consumers.
BUG=webrtc:7177
NOTRY=True
Review-Url: https://codereview.webrtc.org/2695203004
Cr-Commit-Position: refs/heads/master@{#16740}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2688943003
Cr-Commit-Position: refs/heads/master@{#16652}
Create the RTCRtpReceiverDelegate which is a wrapper over
webrtc::RtpReceiverObserverInterface.
The callback will be called whenever the first rtp packet is received.
Related CL: https://codereview.webrtc.org/2531333003/
BUG=webrtc:6742
Review-Url: https://codereview.webrtc.org/2641923003
Cr-Commit-Position: refs/heads/master@{#16501}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
A decision has been made to not use prefix header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2590823002
Cr-Commit-Position: refs/heads/master@{#16056}
H264 High profile is causing problems for some downstream apps. Therefore,
put it behind a field trial instead.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2580963004
Cr-Commit-Position: refs/heads/master@{#15663}
This CL makes a method available on the AVFoundationVideoCapturer
that adapts the output format of captured video to the specified
width and height.
BUG=webrtc:6753
Review-Url: https://codereview.webrtc.org/2528493004
Cr-Commit-Position: refs/heads/master@{#15351}
Reason for revert:
Relanding after known downstream breakages have been fixed.
Original issue's description:
> Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
>
> Reason for revert:
> Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio
>
> Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.
>
> Original issue's description:
> > Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
> >
> > Replaced with webrtc::VideoFrame.
> >
> > TBR=mflodman@webrtc.org
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> > Cr-Commit-Position: refs/heads/master@{#14885}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/7341ab8e2505c9763d208e069bda269018357e7d
> Cr-Commit-Position: refs/heads/master@{#14886}
TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2487633002
Cr-Commit-Position: refs/heads/master@{#15039}
Reason for revert:
Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio
Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.
Original issue's description:
> Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
>
> Replaced with webrtc::VideoFrame.
>
> TBR=mflodman@webrtc.org
> BUG=webrtc:5682
>
> Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> Cr-Commit-Position: refs/heads/master@{#14885}
TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2471783002
Cr-Commit-Position: refs/heads/master@{#14886}
This is to avoid a naming conflict with webrtc::RTCStatsReport that is
surfaced if you try to include it in peerconnectioninterface.h.
Background: The current stats is very much non-spec-compliant. A new
stats collection API is underway that is meant to be spec-compliant.
Some classes in Chromium and webrtc/sdk/objc have spec-compliant names
but non-spec-compliant behavior. These are being renamed to "Legacy" so
that new spec-compliant classes can be added with the correct names.
BUG=chromium:627816
TBR=tkchin@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2313943002
Cr-Commit-Position: refs/heads/master@{#14150}
This CL handles frame rotation by updating the OpenGL vertex data in
RTCOpenGLVideoRenderer, instead of calling the expensive
libyuv::I420Rotate that will rotate the actual memory. Also, we can
handle rotated native frames instead of falling back to
NativeToI420Buffer.
Review-Url: https://codereview.webrtc.org/2176623002
Cr-Commit-Position: refs/heads/master@{#13715}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Fix for RtcEventLog ObjC interface
>
> This moves the RtcEventLog start/stop functions to PeerConnection on the objC interface.
>
> BUG=
> R=tkchin@webrtc.org
>
> Committed: c43bf56ef1TBR=tkchin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2110373002
Cr-Commit-Position: refs/heads/master@{#13338}
