The check: 'RTC_CHECK_GE(slice_height, height);' has been observed to
fail after a reconfig. It looks like |slice_height| is still using the
previous resolution. |slice_height| isn't used for texture output and
hopefully this issue is texture specific. This CL only extracts and
checks |slice_height| when it's actually used.
BUG=b/35932686
Review-Url: https://codereview.webrtc.org/2736603003
Cr-Commit-Position: refs/heads/master@{#17065}
To properly test the functionality, following changes were needed
- Make RTCMTLVideoView compiliable for all cpu architectures not just arm64.
This is needed so that the test can run on any device and on simulator as well.
- Refactor RTCMTLVideoView to have mockable class methods.
The unittest class, RTCMTLVideoViewTests was designed to provide easy transition
to XCTest when the time comes for that.
To transition to XCTest it would suffice to inherit from XCTestCase and remove
the gtest methods.
BUG=webrtc:7079
Review-Url: https://codereview.webrtc.org/2723903003
Cr-Commit-Position: refs/heads/master@{#17014}
- 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}
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}
When textures are not enabled and we are using byte buffer outputs, the
decoder is currently crashing for odd heights because of an RTC_CHECK.
This CL removes the check and handles the pointer offset to the chroma
planes for the odd height case instead.
This has been verified to work correctly on a Pixel device.
BUG=webrtc:6651
Review-Url: https://codereview.webrtc.org/2709923005
Cr-Commit-Position: refs/heads/master@{#16805}
After this change, all calls to MediaCodecVideoEncoder must be made on
the same task queue. Removes OnCodecThread suffix from methods since it
is no longer meaningful.
BUG=webrtc:6290
Review-Url: https://codereview.webrtc.org/2669093004
Cr-Commit-Position: refs/heads/master@{#16792}
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}
Moves CameraCapturer, CameraSession, Camera1Session and Camera2Session
away from the public API.
BUG=webrtc:7172
Review-Url: https://codereview.webrtc.org/2699713004
Cr-Commit-Position: refs/heads/master@{#16723}
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/2690593002
Cr-Commit-Position: refs/heads/master@{#16675}
This is necessary in case the drawer doesn't cover all the pixels.
BUG=None
Review-Url: https://codereview.webrtc.org/2704663002
Cr-Commit-Position: refs/heads/master@{#16671}
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}
Previously, was only checking the Android SDK version. But it also needs
to check for the presence of the connectivity manager service.
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2697943002
Cr-Commit-Position: refs/heads/master@{#16631}
Reason for revert:
Breaks AppRTCMobile interoperability. The ICE candidate URL shouldn't be signaled between endpoints, it's only there for informational purposes.
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> 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/2690593002
> Cr-Original-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16615}
> Committed: 45efce01c7TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2699533002
Cr-Commit-Position: refs/heads/master@{#16616}
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/2690593002
Cr-Original-Commit-Position: refs/heads/master@{#16593}
Committed: 8586c8ee88
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16615}
If android_setsocknetwork() is available, and it fails, then bind()
should *not* be called, and an error should be returned.
If it succeeds, then bind should be called, but with an "any" address.
This is to prevent cases where sockets are sent with a source address
that doesn't match the network interface they're sent on. See bug below.
This CL also changes "NetworkBinderResults" to an enum class, and
renames it to "NetworkBinderResult".
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2646863005
Cr-Commit-Position: refs/heads/master@{#16597}
To ensure compliance with older version high profile should appear in local SDP
before baseline profile.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2696733002
Cr-Commit-Position: refs/heads/master@{#16596}
Reason for revert:
Breaks downstream application's build
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> 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/2690593002
> Cr-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2692993002
Cr-Commit-Position: refs/heads/master@{#16595}
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/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}
This helps show where ownership is transfered between objects.
Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.
BUG=None
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
Currently filed trial value which controls H.264 high profile support is
read once only when factory is created. If field trial value is changed for
the next WebRTC call supported codec list need to be updated as well.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2685183004
Cr-Commit-Position: refs/heads/master@{#16543}
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}
This tag is supposed to be temporary and removed when all Chromium tests
have been migrated to JUnit4.
BUG=webrtc:7123,chromium:640116
NOTRY=True
Review-Url: https://codereview.webrtc.org/2683583002
Cr-Commit-Position: refs/heads/master@{#16465}
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}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Adds ignore for all lint errors in Chromium code. Changes minimum SDK for
instrumentation tests to 16 from 14. Adds TargetApi annotations.
BUG=webrtc:6597
Review-Url: https://codereview.webrtc.org/2670473004
Cr-Commit-Position: refs/heads/master@{#16412}
