Add test for vp8/vp9 qp parser in both videoprocessor_integrationtest.
Check the qp from parser equal to that from the encoder
on every frame in every test.
Add test for vp8/vp9 qp parser in vp8/vp9_impl_test.
Check the qp parser on a single key frame.
BUG=None
Review-Url: https://codereview.webrtc.org/2903163002
Cr-Commit-Position: refs/heads/master@{#18334}
The previous limit leaved no margin for RTT.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2911243002
Cr-Commit-Position: refs/heads/master@{#18333}
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.
BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc
Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
Before this CL, the RTP state would be re-randomized after a
recreation of VideoSendStream. That might lead to us sending
a non-compliant RTP stream, which is avoided after the
changes in this CL.
BUG=webrtc:5654
TBR=pbos@webrtc.org # Trivial change to fuzzer.
Review-Url: https://codereview.webrtc.org/2912713002
Cr-Commit-Position: refs/heads/master@{#18322}
This change:
Reduces complexity for audio playout by removing a redundant memcopy in the output audio path.
Adds support for iOS simulator for playout since we now allow the audio layer to ask for different sizes
of audio buffers at each callback. Real iOS devices always asks for the same size, simulators does not.
This change comes without any new cost for real devices.
BUG=b/37580746
Review-Url: https://codereview.webrtc.org/2894873002
Cr-Commit-Position: refs/heads/master@{#18321}
The change is now compatible with the old JVM::Initialize API. The
context is passed to the ContextUtils class when calling its deprecated
signature.
BUG=webrtc:7665
NOTRY=True # Only comment changes since the last patchset.
Review-Url: https://codereview.webrtc.org/2903253004
Cr-Commit-Position: refs/heads/master@{#18268}
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}
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}
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}
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}
- 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}
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}
probing_interval as a name is used for the period that BWE attempt to increase its estimate. The name is confusing since it is not related to "probing" which is a special mechanism for estimating BWE.
BUG=None
Review-Url: https://codereview.webrtc.org/2888893002
Cr-Commit-Position: refs/heads/master@{#18203}
DesktopRect::UnionWith() function has been added by change
https://codereview.webrtc.org/2845213002. This change adds test cases to cover
the newly added logic. More specifically, union between an empty rectangle and a
non-empty one or two empty rectangles.
BUG=webrtc:7541
Review-Url: https://codereview.webrtc.org/2891593003
Cr-Commit-Position: refs/heads/master@{#18201}
This CL adds a log message with the relevant part of the internal state of the echo detector to the text log when this unexpected scenario occurs.
BUG=b/38014838
Review-Url: https://codereview.webrtc.org/2883283002
Cr-Commit-Position: refs/heads/master@{#18185}
