This will be helpful in unittests to EXPECT_EQ reports. It should be a
useful operator to have outside of testing as well.
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2441543002
Cr-Commit-Position: refs/heads/master@{#14767}
or not being collected correctly.
These TODOs are already documented and in greater detail in
rtcstatscollector.cc, but if every discrepency is listed in
rtcstats_objects.h it is easier to get an overview of the progress of
the new GetStats API.
BUG=chromium:627816
TBR=hta@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2443163002
Cr-Commit-Position: refs/heads/master@{#14749}
Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used.
Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.
This cl require some upstream projects to change first.
BUG=webrtc:5332
Review-Url: https://codereview.webrtc.org/2408153002
Cr-Commit-Position: refs/heads/master@{#14733}
Reason for revert:
This was a workaround to help Chrome wire up the googNoiseReduction constraint. However, it was fixed in a different way in Chrome, and this hack is not actually needed.
Original issue's description:
> Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
>
> BUG=chromium:645907
>
> Committed: https://crrev.com/0d14c6abba19295725519ce38105688ae0a62513
> Cr-Commit-Position: refs/heads/master@{#14219}
TBR=pbos@webrtc.org,hta@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:645907
Review-Url: https://codereview.webrtc.org/2433293003
Cr-Commit-Position: refs/heads/master@{#14729}
We need to wait for posted frames to be rendered first in release()
instead of abruptly quitting, in order to simplify testing.
BUG=webrtc:6545
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2440703002 .
Cr-Commit-Position: refs/heads/master@{#14722}
The stat is currently always set to zero until the residual echo detector has landed.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2431443003
Cr-Commit-Position: refs/heads/master@{#14721}
Simplify the AudioMixer::Source interface and update the mixer
implementation to the new interface.
Instead of asking a mixer source to provide a pointer to an AudioFrame
during each mixing iteration, a mixer should supply a pointer to its
own AudioFrame.
This simplifies lifetime issues as sources do not give away an
internal pointer.
Implementation: when an audio source is added, the mixer allocates a
new AudioFrame. The audio frame is kept together in the internal class
SourceStatus together with the audio source pointer until the source
is removed.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2420913002
Cr-Commit-Position: refs/heads/master@{#14713}
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.
This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.
It will also create less build dependencies when the new mixer has replaced the old one.
NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
This class is logically parallel with the {Audio,Video}ReceiveStream
classes. Its purpose is to describe a receive stream of FlexFEC packets,
through the corresponding config.
Functionally, this class simply forwards the received RTP packets
to its FlexfecReceiver, which returns recovered packets to the
Call level, for appropriate demultiplexing based on SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2397843005
Cr-Commit-Position: refs/heads/master@{#14704}
This CL changes YuvConverter to use an OpenGL Framebuffer as rendering
target instead of an EGL pixel buffer surface. The purpose is to reduce
the number of EGL contexts and to be able to use YuvConverter from
EglRenderer without having to detach the EGL surface.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2436653003
Cr-Commit-Position: refs/heads/master@{#14699}
The purpose is to prepare for a TextureViewRenderer that will need the
new functionality.
The new functionality is:
* Be able to create an EglRenderer using a SurfaceTexture.
* Fps reduction logic.
* Log statistics every 4 seconds regardless of framerate.
* Include swap buffer time in statistics.
* Use EglBase10 if texture frames are disabled.
* Function for printing stack trace of render thread.
* Public clearImage() function for clearing the EGLSurface.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2428933002
Cr-Commit-Position: refs/heads/master@{#14698}
An audio track with a level controller with the correct initialization
value can be created by a combination of
PeerConnectionFactory::CreateAudioTrack(..., audio_source) and
either
audio_source = PeerConnectionFactory::CreateAudioSource(constraints) or
audio_source = PeerConnectionFactory::CreateAudioSource(audio_options).
NOTRY=True
BUG=webrtc:6386
Review-Url: https://codereview.webrtc.org/2408143003
Cr-Commit-Position: refs/heads/master@{#14693}
Reason for revert:
Breaks internal project
Original issue's description:
> Move current bitstream parser to more appropriate directory.
>
> This CL groups together the code that has to do with parsing H264 bitstreams.
> This code logically belongs together, and having it in the same directory not
> only simplifies things from a project structure perspective, but also makes it
> easier to refactor out common parts incrementally.
> An added benefit is that this simplifies modular compilation, where for example
> one would like a build of WebRTC without the H264 codec-specific parts.
>
> BUG=webrtc:6338
>
> Committed: https://crrev.com/cc6817e9ce4a5ffc73efb660cf0368afbc7d9a4f
> Cr-Commit-Position: refs/heads/master@{#14684}
TBR=magjed@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:6338
Review-Url: https://codereview.webrtc.org/2430353004
Cr-Commit-Position: refs/heads/master@{#14685}
This CL groups together the code that has to do with parsing H264 bitstreams.
This code logically belongs together, and having it in the same directory not
only simplifies things from a project structure perspective, but also makes it
easier to refactor out common parts incrementally.
An added benefit is that this simplifies modular compilation, where for example
one would like a build of WebRTC without the H264 codec-specific parts.
BUG=webrtc:6338
Review-Url: https://codereview.webrtc.org/2370853005
Cr-Commit-Position: refs/heads/master@{#14684}
YuvConverter is complex class that deserves its own file. It is also used outside of SurfaceTextureHelper.
BUG=webrtc:6470
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2426023002 .
Cr-Commit-Position: refs/heads/master@{#14683}
This makes it possible for external applications to use this class.
BUG=webrtc:6524
NOTRY=True
Review-Url: https://codereview.webrtc.org/2430693002
Cr-Commit-Position: refs/heads/master@{#14679}
This cl now makes cricket::VideoFrame and cricket::WebRtcVideoFrame aliases for webrtc::VideoFrame.
Reason for revert:
Fixing backwards compatibility issues.
Original issue's description:
> Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ )
>
> Reason for revert:
> Breaks compile for Chromium builds:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/10761
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/18142
>
> FAILED: obj/remoting/protocol/protocol/webrtc_video_renderer_adapter.o
> ../../remoting/protocol/webrtc_video_renderer_adapter.cc:110:52: error: no member named 'transport_frame_id' in 'cricket::VideoFrame'
> weak_factory_.GetWeakPtr(), frame.transport_frame_id(),
> ~~~~~ ^
> 1 error generated.
>
> Please run chromium trybots as described at https://webrtc.org/contributing/#tryjobs-on-chromium-trybots before relanding.
>
> Original issue's description:
> > Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
> > all methods but a few constructors. And similarly for the
> > subclass cricket::WebRtcVideoFrame.
> >
> > TBR=tkchin@webrtc.org # Added an include line
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/dda6ec008a0fc8d52e118814fb779032e8931968
> > Cr-Commit-Position: refs/heads/master@{#14576}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,nisse@webrtc.org
> NOTRY=True
> NOPRESUBMIT=True
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d36dd499c8f253cbcf37364c2a070c2e8c7100e9
> Cr-Commit-Position: refs/heads/master@{#14583}
TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2411953002
Cr-Commit-Position: refs/heads/master@{#14678}
Reason for revert:
Breaks internal project.
Original issue's description:
> Support for video file instead of camera and output video out to file
>
> When video out to file is enabled the remote video which is recorded is
> not show on screen.
>
> You can use this command line for file input and output:
> monkeyrunner ./webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py --devname 02157df28cd47001 --videoin /storage/emulated/0/reference_video_1280x720_30fps.y4m --videoout /storage/emulated/0/output.y4m --videoout_width 1280 --videoout_height 720 --videooutsave /tmp/out.y4m
>
> BUG=webrtc:6545
>
> Committed: https://crrev.com/44666997ca912705f8f96c9bd211e719525a3ccc
> Cr-Commit-Position: refs/heads/master@{#14660}
TBR=magjed@webrtc.org,sakal@webrtc.org,jansson@chromium.org,mandermo@google.com,mandermo@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2425763003
Cr-Commit-Position: refs/heads/master@{#14664}
When video out to file is enabled the remote video which is recorded is
not show on screen.
You can use this command line for file input and output:
monkeyrunner ./webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py --devname 02157df28cd47001 --videoin /storage/emulated/0/reference_video_1280x720_30fps.y4m --videoout /storage/emulated/0/output.y4m --videoout_width 1280 --videoout_height 720 --videooutsave /tmp/out.y4m
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2273573003
Cr-Commit-Position: refs/heads/master@{#14660}
The default implementations are provided as to not break Chromium mocks,
as soon as we have done a successful roll they should be updated and the
default implementations removed.
TBR=hta@webrtc.org, deadbeef@webrtc.org
NOTRY=True
BUG=chromium:654927
Review-Url: https://codereview.webrtc.org/2414613003
Cr-Commit-Position: refs/heads/master@{#14617}
- Rename the data codec payload types to end with "PlType" instead of "Id", for consistency.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2397413002
Cr-Commit-Position: refs/heads/master@{#14581}
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.
This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).
BUG=webrtc:6393
Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
The purpose is to prepare for a TextureViewRenderer that will share the
EGL rendering code.
Two functional changes are also included:
* The implementation of SurfaceHolder.Callback.surfaceDestroyed will now
block until the EGL surface is released. This is done in order to
comply with the documentation that says: "If you have a rendering
thread that directly accesses the surface, you must ensure that thread
is no longer touching the Surface before returning from this function."
* We will no longer try to hide render glitches during layout changes.
This was a lost cause anyway.
BUG=webrtc:6407
Review-Url: https://codereview.webrtc.org/2399463006
Cr-Commit-Position: refs/heads/master@{#14570}
The RTCStatsCollector collects candidates from candidate pairs. Note
that there may be other candidates that are not paired with anything,
stats for these should also be produced before closing crbug.com/632723.
[1] https://w3c.github.io/webrtc-stats/#icecandidate-dict*
BUG=chromium:627816, chromium:632723
Review-Url: https://codereview.webrtc.org/2384143002
Cr-Commit-Position: refs/heads/master@{#14565}
This should allow us to enable Intel HW VP8 encoder again.
BUG=webrtc:6232,b/30947951
Review-Url: https://codereview.webrtc.org/2263043003
Cr-Commit-Position: refs/heads/master@{#14552}
The warning previously suppressed made it possible to define tings like
constructors in the header, and "complex" objects did not need to have
an explicit out-of-line copy constructor, destructor, etc.
To be able to not suppress this warning, the RTCStats macro was split
into a WEBRTC_RTCSTATS_DECL() and WEBRTC_RTCSTATS_IMPL() for .h and .cc
respectively. Some copy constructors are also defined.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2373503002
Cr-Commit-Position: refs/heads/master@{#14545}
This error occurred if you created a data channel before there's a data
m= section. But this is expected to happen, since creating a data
channel is how you get a data m= section in the first place.
BUG=chromium:579430
NOTRY=true
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2396013002
Cr-Commit-Position: refs/heads/master@{#14537}
RTCStatsMemberInterface::Type's kBool and kSequenceBool.
This means that RTCStats-derived classes ("RTCStats-derived
dictionaries"[1]) can contain boolean and sequence of boolean members.
[1] https://w3c.github.io/webrtc-stats/
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2387343002
Cr-Commit-Position: refs/heads/master@{#14509}
hbos and hta are already OWNERS of webrtc/stats/ and of rtcstats* files
(per-file rtcstats*=) in webrtc/api/. When the webrtc/api/stats/ folder
was created we forgot to add this OWNERS file (per-file OWNERS does not
apply to subfolders apparently).
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2392633002
Cr-Commit-Position: refs/heads/master@{#14482}
