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}
Since the RtpStreamId and RepairedRtpStreamId extensions can have variable
length, it makes no sense for them to have a constant valueSize field.
The header length calculation in RtpHeaderExtensionMap needed to be changed
for this because it previously worked with the assumption that all header
types have a constant size. Now it's the caller's job to specify the length
of the extensions that it might use.
BUG=webrtc:7433
Review-Url: https://codereview.webrtc.org/2867713003
Cr-Commit-Position: refs/heads/master@{#18179}
This small piece of logic is duplicated in DxgiDuplicatorController,
DxgiAdapterDuplicator and desktop_configuration.mm. Meanwhile, the
implementation in desktop_configuration.mm is not safe. So I think a function in
DesktopRect to cover the requirement could be more efficient and safer.
BUG=webrtc:7541
Review-Url: https://codereview.webrtc.org/2845213002
Cr-Commit-Position: refs/heads/master@{#18171}
Reason for revert:
Breaking downstream projects.
Original issue's description:
> Split iOS sdk in to separate targets
>
> This CL splits the iOS sdk into separate static libraries for video,
> audio, ui, common, and peerconnection-related code. This will in the
> future make it easier to compile WebRTC without unneeded components.
>
> BUG=webrtc:4867
>
> Review-Url: https://codereview.webrtc.org/2862543002
> Cr-Commit-Position: refs/heads/master@{#18166}
> Committed: 52c83fe710TBR=magjed@webrtc.org,denicija@webrtc.org,tkchin@webrtc.org,henrika@webrtc.org,kthelgason@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4867
Review-Url: https://codereview.webrtc.org/2890513002
Cr-Commit-Position: refs/heads/master@{#18170}
This CL changes the updating of a loop index from using the modulus
operator to using a conditional, avoiding the divisions done in the
modulus operator which had a measurable impact on mpbile
platforms.
Additionally, there is a similar removal of another modulus operator, but the impact of that should be negligible.
BUG=webrtc:7666
Review-Url: https://codereview.webrtc.org/2882183004
Cr-Commit-Position: refs/heads/master@{#18168}
Negating an int can result in a value that cannot be represented as an int. This is fixed here by using a 64 bit variable.
BUG=chromium:663611
Review-Url: https://codereview.webrtc.org/2879863002
Cr-Commit-Position: refs/heads/master@{#18167}
This CL splits the iOS sdk into separate static libraries for video,
audio, ui, common, and peerconnection-related code. This will in the
future make it easier to compile WebRTC without unneeded components.
BUG=webrtc:4867
Review-Url: https://codereview.webrtc.org/2862543002
Cr-Commit-Position: refs/heads/master@{#18166}
This is a robustness test for the residual echo detector, that can help to detect numerical issues.
BUG=b/38014838
Review-Url: https://codereview.webrtc.org/2877803002
Cr-Commit-Position: refs/heads/master@{#18165}
This CL removes the residual echo detector from the list of
modules in APM that requires band-splitting.
BUG=webrtc:6220, webrtc:6183
Review-Url: https://codereview.webrtc.org/2884913002
Cr-Commit-Position: refs/heads/master@{#18164}
This header was not tracked by a GN target and in the discussion on
https://bugs.chromium.org/p/webrtc/issues/detail?id=7617 we decided
to also move it under webrtc/base.
I checked in chromium code search and it seems safe to move it
without creating a stub header in webrtc/system_wrappers.
BUG=webrtc:7617
Review-Url: https://codereview.webrtc.org/2882673002
Cr-Commit-Position: refs/heads/master@{#18151}
This CL moves the residual echo detector to reside outside of
the band-scheme in APM. The benefit of this is that the
residual echo detector will then no longer enforce the
band-splitting to be used when it is the only active component
inside APM.
This CL also introduces diagnostic dumping of data inside the
residual echo detector.
BUG=webrtc:6220, webrtc:6183
Review-Url: https://codereview.webrtc.org/2884593002
Cr-Commit-Position: refs/heads/master@{#18150}
To make the distinction for stats, add a |recovered| flag to
RtpPacketReceived.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2693123002
Cr-Commit-Position: refs/heads/master@{#18103}
The audio mixer has a subcomponent called FrameCombiner, which uses an
AudioProcessing instance as a limiter. The limiter smoothly increases
the volume to avoid causing clipping.
The limiter was created in a default configuration causing the
ResidualEchoDetector submodule of AudioProcessing to be
activated. That submodule operates in the band-split domain (see
AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive()).
There is a goal to remove the (expensive and unnecessary)
band-splitting from AudioMixer. This change helps accomplish that. (It
can't be done yet, because the actual limiter sub-component of APM
also operates in the band-split domain).
BUG=webrtc:6185
Review-Url: https://codereview.webrtc.org/2875623002
Cr-Commit-Position: refs/heads/master@{#18090}
In the windows audio device implementation, play block size should be
number of samples per 10ms times the number of channels. Meanwhile
RequestPlayoutData is expecting number of samples per channel in a
block, and we should pass in the per channel number here to avoid debug
check.
BUG=7627
Review-Url: https://codereview.webrtc.org/2876593003
Cr-Commit-Position: refs/heads/master@{#18088}
RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.
A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.
With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.
BUG=webrtc:7610, webrtc:7625
Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}
