Reason for revert:
Failed on Win 10 Chrome FYI.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#
WebRtcBrowserTest
#
Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}
TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
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}
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}
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 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}
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}
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}
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}
This change allows the application to limit the bitrate of the outgoing
audio and video streams at runtime. The API roughly follows the WebRTC
API draft, defining the RTCRtpParameters structure witn exactly one
encoding (simulcast streams are not exposed in the API for now).
(https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters)
BUG=
Review URL: https://codereview.webrtc.org/1788583004
Cr-Commit-Position: refs/heads/master@{#12025}
This cl copies the value of cricket::VideoCapturer::IsScreencast into
a flag in VideoOptions. It is passed on via the chain
VideortpSender::SetVideoSend
WebRtcVideoChannel2::SetVideoSend
WebRtcVideoChannel2::SetOptions
WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions
Where it's used, in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up
in parameters_, instead of calling capturer_->IsScreencast().
Doesn't touch screencast logic related to cpu adaptation, since that
code is in flux in a different cl.
Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options.
In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests.
Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters.
BUG=webrtc:5426
R=pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1711763003 .
Cr-Commit-Position: refs/heads/master@{#11837}
Rename SetCodecAndOptions to SetCodec, it no longer sets or uses the
VideoOptions. In MediaConfig, collect the video-related flags into a
struct.
As a followup, it should be possible to delete VideoOptions from
VideoSendParameters and VideoSendStreamParameters.
TBR=pthatcher@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1745003002
Cr-Commit-Position: refs/heads/master@{#11828}
The purose of this cl is to remove dependency on cricket::VideoCapturer from WebRtcVideoChannel2.
This cl change CPU adaptation to use a new VideoSinkWants.Resolution
Cl is WIP and uploaded to start the discussion.
Tested on a N5 with hw acceleration turned off.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1695263002
Cr-Commit-Position: refs/heads/master@{#11804}
For audio, the flag is apparently unused. For video, the flag is moved to
VideoSendParameters, with the intention to keep only per-stream flags in
VideoOptions. The flag is used for the webrtcvideoengine2 logic commented like
// Conference mode screencast uses 2 temporal layers split at 100kbit.
// For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
// on the VideoCodec struct as target and max bitrates, respectively.
// See eg. webrtc::VP8EncoderImpl::SetRates().
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1697163002
Cr-Commit-Position: refs/heads/master@{#11651}
