Passing transport_frame_id() to VideoSink will allow to identify incoming video
frames, which will make it possible to correlate video frames on the
sender and on the receiver.
BUG=chromium:621691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2088953002 .
Cr-Commit-Position: refs/heads/master@{#13596}
VideoDecoderParams contains the id of the receive video
stream. Motivation behind this change is to enable down
stream apps easier map raw non-decoded data to incoming
streams.
BUG=b/28636393
Review-Url: https://codereview.webrtc.org/2052233002
Cr-Commit-Position: refs/heads/master@{#13250}
Don't use VideoFrameBuffer::MutableDataY and friends, instead, use
I420Buffer::SetToBlack.
Also introduce static method I420Buffer::Create, to create an object and
return a scoped_refptr.
TBR=marpan@webrtc.org # Trivial change to video_denoiser.cc
BUG=webrtc:5921
Review-Url: https://codereview.webrtc.org/2078943002
Cr-Commit-Position: refs/heads/master@{#13212}
This change reduces the number of times the Android hardware video
encoder is reconfigured when making an outgoing call. With this change,
the encoder should only be initialized once as opposed to the ~3 times
it happens currently.
Before the fix, the following sequence of events caused the extra
reconfigurations:
1. After the SetLocalDescription call, the WebRtcVideoSendStream is created.
All frames from the camera are dropped until the corresponding
VideoSendStream is created.
2. SetRemoteDescription() triggers the VideoSendStream creation. At
this point, the encoder is configured for the first time, with the
frame dimensions set to a low resolution default (176x144).
3. When the first video frame is received from the camera after the
VideoSendStreamIsCreated, the encoder is reconfigured to the correct
dimensions. If we are using the Android hardware encoder, the default
configuration is set to encode from a memory buffer (use_surface=false).
4. When the frame is passed down to the encoder in
androidmediaencoder_jni.cc EncodeOnCodecThread(), it may be stored in
a texture instead of a memory buffer. In this case, yet another
reconfiguration takes place to enable encoding from a texture.
5. Even if the resolution and texture flag were known at the start of
the call, there would be a reconfiguration involved if the camera is
rotated (such as when making a call from a phone in portrait orientation).
The reason for that is that at construction time, WebRtcVideoEngine2
sets the VideoSinkWants structure parameter to request frames rotated
by the source; the early frames will then arrive in portrait resolution.
When the remote description is finally set, if the rotation RTP extension
is supported by the remote receiver, the source is asked to provide
non-rotated frames. The very next frame will then arrive in landscape
resolution with a non-zero rotation value to be applied by the receiver.
Since the encoder was configured with the last (portrait) frame size,
it's going to need to be reconfigured again.
The fix makes the following changes:
1. WebRtcVideoSendStream::OnFrame() now caches the last seen frame
dimensions, and whether the frame was stored in a texture.
2. When the encoder is configured the first time
(WebRtcVideoSendStream::SetCodec()) - the last seen frame dimensions
are used instead of the default dimensions.
3. A flag that indicates if encoding is to be done from a texture has
been added to the webrtc::VideoStream and webrtc::VideoCodec structs,
and it's been wired up to be passed down all the way to the JNI code in
androidmediaencoder_jni.cc.
4. MediaCodecVideoEncoder::InitEncode is now reading the is_surface
flag from the VideoCodec structure instead of guessing the default as
false. This way we end up with the correct encoder configuration the
first time around.
5. WebRtcVideoSendStream now takes an optimistic guess and requests non-
rotated frames when the supported RtpExtensions list is not available.
This makes the "early" frames arrive non-rotated, and the cached dimensions
will be correct for the common case when the rotation extension is supported.
If the other side is an older endpoint which does not support rotation,
the encoder will have to be reconfigured - but it's better to penalize the
uncommon case rather than the common one.
Review-Url: https://codereview.webrtc.org/2067103002
Cr-Commit-Position: refs/heads/master@{#13173}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
Instead of the default copy constructor, the Copy() method has to be used. In this CL, the number of copies has been reduced significantly in production code. One case in the video engine remains, where we need to restart a video stream. Even in that case, I'm sure we could avoid it, but for this particular CL, I decided against it to keep things simple (and it's also an edge case). Most importantly, creating copies is made harder and the interface encourages ownership transfers.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2042603002 .
Cr-Commit-Position: refs/heads/master@{#13102}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
This means there's only one thread hop to the worker thread.
At the video engine level, SetOptions and SetSource
are combined into one method (all within the same critical section)
which ensures that no frame will be encoded while SetVideoSend
is only partially finished.
BUG=webrtc:5691
Review-Url: https://codereview.webrtc.org/1838413002
Cr-Commit-Position: refs/heads/master@{#13022}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
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}
Reason for revert:
Speculative revert to see if failures on the DrMemory bot are related to this cl. See e.g. here:
https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243
UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
# 0 CopyRow_AVX
# 1 CopyPlane
# 2 I420Copy
# 3 webrtc::ExtractBuffer
# 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
# 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
# 6 FakeWebRtcVideoCaptureModule::SendFrame
# 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
# 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>
Original issue's description:
> Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
>
> Reason for revert:
> I plan to reland this change in a week or two, after downstream users are updated.
>
> Original issue's description:
> > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> >
> > Reason for revert:
> > Breaks chrome FYI bots.
> >
> > Original issue's description:
> > > Delete webrtc::VideoFrame methods buffer and stride.
> > >
> > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > to not imply an AddRef.
> > >
> > > BUG=webrtc:5682
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > Cr-Commit-Position: refs/heads/master@{#12558}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> Cr-Commit-Position: refs/heads/master@{#12721}
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@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/1983583002
Cr-Commit-Position: refs/heads/master@{#12745}
Wires up existing libvpx_build_vp9==0 GYP flag into WebRTC and makes VP9
optional. Change is GYP only for now since libvpx's GN files build VP9
unconditionally.
BUG=webrtc:5884
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1970343002 .
Cr-Commit-Position: refs/heads/master@{#12741}
Reason for revert:
Breaks chrome FYI bots.
Original issue's description:
> Delete webrtc::VideoFrame methods buffer and stride.
>
> To make the HasOneRef/IsMutable hack work, also had to change the
> video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> to not imply an AddRef.
>
> BUG=webrtc:5682
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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/1935443002
Cr-Commit-Position: refs/heads/master@{#12558}
To make the HasOneRef/IsMutable hack work, also had to change the
video_frame_buffer method to return a const ref to a scoped_ref_ptr,
to not imply an AddRef.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1900673002
Cr-Commit-Position: refs/heads/master@{#12557}
This fixes a problem where "b=AS" and "x-google-start-bitrate" can't
be used together. It also starts taking the minimum of "b=AS" and
"x-google-max-bitrate", instead of just letting "b=AS" win.
BUG=webrtc:5811
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1904063003 .
Cr-Commit-Position: refs/heads/master@{#12519}
This field only existed as an implementation detail for getting the
codecs sorted, so it doesn't need to be in the public interface.
It cluttered the code and undesirably affected codec comparisons,
causing the video encoder to be reconfigured if a codec's preference
changed but nothing else did.
BUG=webrtc:5690
Review URL: https://codereview.webrtc.org/1845673002
Cr-Commit-Position: refs/heads/master@{#12349}
This CL generates FMTP parameters that allow H.264 interoperation
with Firefox for the default codec list.
BUG=chromium:591971
Review URL: https://codereview.webrtc.org/1880963002
Cr-Commit-Position: refs/heads/master@{#12333}
Unit tests are updated to test that screen share is not adapted but it does not change the VideoSinkWants in WebRtcVideoEngine2::SendStream due to a switch to screen share. The reason is that it works anyway and sprang is looking into how to do adaptation based on frame rate as well and use the adapter for screen share as well.
BUG=webrtc:5688, webrtc:5426
R=nisse@webrtc.org, pbos@webrtc.org, sprang@google.com
Review URL: https://codereview.webrtc.org/1836043004 .
Cr-Commit-Position: refs/heads/master@{#12240}
To replace the SmoothsRenderedFrames method, added a corresponding
flag to VideoReceiveStream::Config instead.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1818023002
Cr-Commit-Position: refs/heads/master@{#12102}
This will allow a sender to stop/start sending media on the
application's demand.
Among other things, this can allow an application to set a track on a
sender while the encoding(s) are inactive, allowing the encoder to be
initialized for that track, then later set the encodings to "active"
to instantly start sending the track.
Review URL: https://codereview.webrtc.org/1822923002
Cr-Commit-Position: refs/heads/master@{#12094}
This change enables voice-only calls to keep track of the network state.
This is only a partial fix - the last modality to change state controls
the state for the entire call, so a call with a failed video transport
will also stop sending audio packets. Handling this condition correctly
would require the call to keep track of network state for each media
type separately, and take care of conditions such as a failed video
channel getting removed, while a functioning audio channel remains.
BUG=webrtc:5307
Review URL: https://codereview.webrtc.org/1757683002
Cr-Commit-Position: refs/heads/master@{#12093}
webrtc::VideoRenderer class, replacing it by rtc::VideoSinkInterface.
The next step is to convert all places where a renderer is attached to
rtc::VideoSourceInterface, and at that point, the
SmoothsRenderedFrames method can be replaced by a flag
rtc::VideoSinkWants::smoothed_frames.
Delete unused method IsTextureSupported.
Delete unused time argument to RenderFrame.
Let webrtc::VideoRenderer inherit rtc::VideoSinkInterface. Rename RenderFrame --> OnFrame.
TBR=kjellander@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1814763002
Cr-Commit-Position: refs/heads/master@{#12070}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1823503002
Cr-Commit-Position: refs/heads/master@{#12062}
Reason for revert:
I'm really sorry for having to revert this but it seems this hit an unexpected compile error downstream:
webrtc/media/sctp/sctpdataengine.cc: In function 'void cricket::VerboseLogPacket(const void*, size_t, int)':
webrtc/media/sctp/sctpdataengine.cc:172:37: error: invalid conversion from 'const void*' to 'void*' [-fpermissive]
data, length, direction)) != NULL) {
^
In file included from webrtc/media/sctp/sctpdataengine.cc:20:0:
third_party/usrsctp/usrsctplib/usrsctp.h:964:1: error: initializing argument 1 of 'char* usrsctp_dumppacket(void*, size_t, int)' [-fpermissive]
usrsctp_dumppacket(void *, size_t, int);
^
I'm sure you can fix this easily and just re-land this CL, while I'm going to look into how to add this warning at the public bots (on Monday).
Original issue's description:
> Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
>
> This CL removes copy and assign support from Buffer and changes various
> parameters from Buffer to CopyOnWriteBuffer so they can be passed along
> and copied without actually copying the underlying data.
>
> With this changed some parameters to be "const" and fixed an issue when
> creating a CopyOnWriteBuffer with empty data.
>
> BUG=webrtc:5155
>
> Committed: https://crrev.com/944c39006f1c52aee20919676002dac7a42b1c05
> Cr-Commit-Position: refs/heads/master@{#12058}
TBR=kwiberg@webrtc.org,tkchin@webrtc.org,tommi@webrtc.org,pthatcher@webrtc.org,jbauch@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1817753003
Cr-Commit-Position: refs/heads/master@{#12060}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1785713005
Cr-Commit-Position: refs/heads/master@{#12058}
