A low priority task queue is added to WebRTCVoiceEngine. The
start/stop debug calls make file logging happen on the task queue.
In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
so that it can be shared for low priority tasks between different
subcomponents. It will require some changes to MediaEngine,
CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
projects.
A task queue must be created and destroyed from the same thread. With
this CL that will be the worker thread, which creates and destroys
WebRTCVoiceEngine. With the dependent CL, it will probably change to
the signaling thread.
NOTRY=True # tests just passed
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2896813002
Cr-Commit-Position: refs/heads/master@{#18252}
Reason for revert:
Causes a new TSan race warning. Will reland after fixing. Note this is the same race as will be fixed by https://codereview.webrtc.org/2876273002/.
Original issue's description:
> Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations.
>
> The deadlock occurs if the AsyncInvoker is destroyed on thread A while
> a task on thread B is running, which AsyncInvokes a task back on thread
> A.
>
> This was causing pending_invocations_ to end up negative, because
> an AsyncClosure that's never added to a thread's message queue (due to
> the "destroying_" flag) caused the count to be decremented but not
> incremented.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2885143006
> Cr-Commit-Position: refs/heads/master@{#18225}
> Committed: ef37ca5fb3TBR=nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2904543003
Cr-Commit-Position: refs/heads/master@{#18248}
Change packetizer interface to explicitly return number of packets
instead of a last flag. Account for extra space needed in the last
packet.
BUG=webrtc:7588,webrtc:7594
Review-Url: https://codereview.webrtc.org/2871173008
Cr-Commit-Position: refs/heads/master@{#18244}
On Linux, during Windwo sharing, the cursore capture may happen in the parent
window of the target. And the parent window may have some decorations added by
window manager(Chrome windows don't have those decorations.), so the relative
cursor position to the parent window with decorations may differ to its child
target window. The offset includes the height of caption bar and the around
shadow and border.
This problem only happens with Window sharing on Linux.
The fix is to translate the coordinates from the parent window to the coordinates space of the target window.
BUG=723889
Review-Url: https://codereview.webrtc.org/2889063002
Cr-Commit-Position: refs/heads/master@{#18243}
This CL implements webrtc::AecDump, which is an interface defined in
https://codereview.webrtc.org/2778783002.
This AudioProcessing submodule writes audio and APM state to a
file. The file writing is done by posting IO tasks
(write_to_file_task.h) on an rtc::TaskQueue. There is an existing
implementation for this through AudioProcessing::StartDebugRecording()
and AudioProcessing::StopDebugRecording(). This implementation still
works, and is used as the default until this dependent CL:
https://codereview.webrtc.org/2896813002/.
To be able to build webrtc without protobuf support, the interface is
isolated from protobuf types. Audio data from AudioProcessing is
passed to AecDumpImpl through the AecDump interface. There it is
stored in protobuf objects, which are posted on the task queue.
This functionality is verified correct by the CL
https://codereview.webrtc.org/2864373002, which enables this recording
submodule in APM tests.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2865113002
Cr-Commit-Position: refs/heads/master@{#18241}
If a frame is dropped and re-encoded because it exceeded the target
bitrate by a large factor, the next frame will be encoded at max qp
(worst quality) in order to get a frame through in a timely manner. The
next frame after this will still have lower quality since the rate
controller essentially gets reset. In order to mitigate that we boost
the qp for that next frame, which brings the stream back to a good
quality quicker.
However, if the network conditions are _really_ bad, this boosted qp
may be too large, causing the frame again to be dropped an re-encoded.
This CL set's a minimum bitrate available in order to enabling the
boosting in the first place.
It also adjusts a timeout (max time between frames in TL0), since a
too small value and very difficult frames in conjunction with the
mentioned bad network could actually cause bad network over-utilization
in turn leading to packet loss and bad follow-on effects to that.
There was also some slop in the rate keeping for the two layers.
This has been tightened up and affected test cases have been fixed.
BUG=webrtc:7694
Review-Url: https://codereview.webrtc.org/2897983002
Cr-Commit-Position: refs/heads/master@{#18236}
configuration, inputs and outputs over a period of time. It is
activated by AudioProcessing::StartRecording. The data is stored in
binary protobuf format in a specified file. The file IO is, as of
this CL, done from the real-time audio thread.
This CL contains an interface for AecDump, a new APM submodule that
will handle the recordings. Calls to the new interface from the
AudioProcessingModule are added. These calls have no effect, and for a
short while, audio_processing_impl.cc will contain two copies of
recording calls.
The original calls are guarded by the WEBRTC_AUDIOPROC_DEBUG_DUMP
preprocessor define. They still have an effect, while the new ones do
not. In the following CLs, the old recording calls will be removed,
and an implementation of AecDump added.
The reasons for the refactoring is to move file IO operations from the
real-time audio thread, to add a top-level low-priority task queue for
logging tasks like this, to simplify and modularize audio_processing_impl.cc
and remove some of the preprocessor directives. These goals will be
archived by the upcoming CLs. The implementation is in
https://codereview.webrtc.org/2865113002.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2778783002
Cr-Commit-Position: refs/heads/master@{#18233}
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.
Also fix a bug in vie_encoder where the codec was not perioducally
updated unless a new bitrate estimate was triggered.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.
BUG=webrtc:7631
Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}
This CL ensures that the output of the level controller is kept
when no other submodules in APM are active
BUG=webrtc:7697,
Review-Url: https://codereview.webrtc.org/2902723002
Cr-Commit-Position: refs/heads/master@{#18230}
This CL increases the transparency of the AEC3 via tuning.
The major changes are
1) Limiting the suppression gain to the 16 bit sample floor.
2) Controlling the rate of the suppression gain increase
according to the signal characteristics.
Apart from these tunings, the code for the suppression gain
was refactored to increase/maintain the code quality after
the above changes.
BUG=webrtc:7519,webrtc:7528, chromium:715893
Review-Url: https://codereview.webrtc.org/2886733002
Cr-Commit-Position: refs/heads/master@{#18229}
Reason for reland:
Chrome encoder implementation fixed.
Original issue's description:
> Revert of Reuse allocated encoders in SimulcastEncoderAdapter. (patchset #15 id:320001 of https://codereview.webrtc.org/2830793005/ )
>
> Reason for revert:
> Breaks Chrome tests.
>
> Original issue's description:
> > Reuse allocated encoders in SimulcastEncoderAdapter.
> >
> > Prior to this change, the SimulcastEncoderAdapter would destroy and create
> > encoders whenever it is being reinitialized. After this change, the
> > SimulcastEncoderAdapter will cache the already allocated encoders, and reuse
> > them after reinitialization.
> >
> > This change will help in reducing the number of PictureID "jumps" that have
> > been seen around encoder reinitialization.
> >
> > TESTED=AppRTCMobile, Chrome desktop, and internal app, with forced encoder reinits every 30 frames and https://codereview.webrtc.org/2833493003/ applied.
> > BUG=webrtc:7475
> >
> > Review-Url: https://codereview.webrtc.org/2830793005
> > Cr-Commit-Position: refs/heads/master@{#18215}
> > Committed: 0b8bfb9d98
>
> TBR=stefan@webrtc.org,noahric@chromium.org,glaznev@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7475
>
> Review-Url: https://codereview.webrtc.org/2893003002
> Cr-Commit-Position: refs/heads/master@{#18216}
> Committed: 56e119e2e8TBR=stefan@webrtc.org,noahric@chromium.org,glaznev@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7475
Review-Url: https://codereview.webrtc.org/2901493002
Cr-Commit-Position: refs/heads/master@{#18228}
When operating on mobile devices, where hardware support is available
for the AEC and NS functionality, it is desirable to be able to
operate without hardcoded behaviors for the WebRTC AGC and HPF.
This CL adds support to allow a field trial to turn these off
whenever that is possible.
BUG=webrtc:6220, webrtc:6183, webrtc:6181
Review-Url: https://codereview.webrtc.org/2876133002
Cr-Commit-Position: refs/heads/master@{#18226}
The deadlock occurs if the AsyncInvoker is destroyed on thread A while
a task on thread B is running, which AsyncInvokes a task back on thread
A.
This was causing pending_invocations_ to end up negative, because
an AsyncClosure that's never added to a thread's message queue (due to
the "destroying_" flag) caused the count to be decremented but not
incremented.
BUG=None
Review-Url: https://codereview.webrtc.org/2885143006
Cr-Commit-Position: refs/heads/master@{#18225}
split domain and only implements floating point operations (to avoid spectral
leakage issues and unnecessary complexity).
The goal of this CL is adding the new sub-module into APM without providing an
implementation that could replace the existing gain control modules. The focus
is in fact on initialization, reset, and configuration of AGC2.
The module itself only applies a hard-coded gain value. This behavior will
change in the coming CLs.
BUG=webrtc:7494
Review-Url: https://codereview.webrtc.org/2848593002
Cr-Commit-Position: refs/heads/master@{#18222}
Packet::GetHeader() wasn't copying playout_delay. As result
playout_delay was ignored when flexfec is enabled.
Patch by Rob McCool <rmccool@google.com>
BUG=webrtc:7590
Review-Url: https://codereview.webrtc.org/2899553003
Cr-Commit-Position: refs/heads/master@{#18218}
The media related stats wasn't working for unsignaled stream because there
is no mapping between the receiver_info and unsignaled tracks.
This CL fixes the issue by adding some special logic to the TrackMediaInfoMap
which would create the mapping.
BUG=b/37836881
BUG=webrtc:7685
TBR=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2883943003
Cr-Commit-Position: refs/heads/master@{#18217}
Reason for revert:
Breaks Chrome tests.
Original issue's description:
> Reuse allocated encoders in SimulcastEncoderAdapter.
>
> Prior to this change, the SimulcastEncoderAdapter would destroy and create
> encoders whenever it is being reinitialized. After this change, the
> SimulcastEncoderAdapter will cache the already allocated encoders, and reuse
> them after reinitialization.
>
> This change will help in reducing the number of PictureID "jumps" that have
> been seen around encoder reinitialization.
>
> TESTED=AppRTCMobile, Chrome desktop, and internal app, with forced encoder reinits every 30 frames and https://codereview.webrtc.org/2833493003/ applied.
> BUG=webrtc:7475
>
> Review-Url: https://codereview.webrtc.org/2830793005
> Cr-Commit-Position: refs/heads/master@{#18215}
> Committed: 0b8bfb9d98TBR=stefan@webrtc.org,noahric@chromium.org,glaznev@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7475
Review-Url: https://codereview.webrtc.org/2893003002
Cr-Commit-Position: refs/heads/master@{#18216}
Prior to this change, the SimulcastEncoderAdapter would destroy and create
encoders whenever it is being reinitialized. After this change, the
SimulcastEncoderAdapter will cache the already allocated encoders, and reuse
them after reinitialization.
This change will help in reducing the number of PictureID "jumps" that have
been seen around encoder reinitialization.
TESTED=AppRTCMobile, Chrome desktop, and internal app, with forced encoder reinits every 30 frames and https://codereview.webrtc.org/2833493003/ applied.
BUG=webrtc:7475
Review-Url: https://codereview.webrtc.org/2830793005
Cr-Commit-Position: refs/heads/master@{#18215}
Instead of only looking at Audio or ANY media type, we ignore duplicate RTCP packets.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2892913002
Cr-Commit-Position: refs/heads/master@{#18213}
This CL reduces the number of VideoSendStream recreations during SDP
renegotiation by checking the FlexFEC field trials before, and not after,
the SDP codec diffing logic.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2882433003
Cr-Commit-Position: refs/heads/master@{#18211}
- Add codec_type-implementation_name label option.
- Update figure title to exclude information that exist in legend.
- Change frame info in title from: # of frames in file -> # of processed frames.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2890223002
Cr-Commit-Position: refs/heads/master@{#18209}
External dependencies need to be updated to call ContextUtils.initialize
before rest of the CL can be landed.
BUG=webrtc:7665
Review-Url: https://codereview.webrtc.org/2893933003
Cr-Commit-Position: refs/heads/master@{#18208}
This CL adjusts the render processing rate such to avoid resampling of the
render signal when that is not needed.
Note that to avoid acquiring more locks than needed, this should be achieved
during initialization.
BUG=webrtc:7667
Review-Url: https://codereview.webrtc.org/2887693002
Cr-Commit-Position: refs/heads/master@{#18207}
