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}
FakeRtpTransportController moves to a common header and its constructor is changed to take a SendSideCongestionController to enable injecting the mock.
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2834663003
Cr-Commit-Position: refs/heads/master@{#18055}
This allows it to be reused with Android AudioRecords created outside
WebRtcAudioRecord. WebRtcAudioEffects provides useful methods for enabling
hardware effects (such as echo cancellation) only on appropriate devices. It
also allows some control of these effects through WebRtcAudioUtils.
BUG=webrtc:7448
Review-Url: https://codereview.webrtc.org/2786603004
Cr-Commit-Position: refs/heads/master@{#18053}
The residual echo likelihood should report a likelihood between 0.0 and 1.0. Currently it can happen that echo likelihoods > 1.0 are reported. As a temporary mitigation to stop this, this CL enforces a hard maximum of 1.0 for the echo likelihood while we investigate the issue further.
BUG=b/38014838
Review-Url: https://codereview.webrtc.org/2861123002
Cr-Commit-Position: refs/heads/master@{#18030}
After a Merge operation, the statistics for number of samples
generated using Expand must be corrected, and the correction can in
fact be negative. However, a bug was introduced in
https://codereview.webrtc.org/1230503003 which uses a size_t to
represent the correction, which leads to wrap-around of the negative
value. This is not a problem in itself, since this value is added to
another size_t, with the effect that the desired subtraction happens
anyway.
The actual problem arises if the statistics are polled/reset before a
subtraction happens -- that is, between an Expand and a Merge
operation. This will lead to an actual wrap-around of the stats value,
and large expand_rate (16384) is reported.
BUG=webrtc:7554
Review-Url: https://codereview.webrtc.org/2859483005
Cr-Commit-Position: refs/heads/master@{#18029}
This is instead derived from pattern_idx inside the frame config.
This also removes active_layer_ use from
ScreenshareLayers::PopulateCodecSpecific and instead ties the layer to
TemporalLayers::FrameConfig.
BUG=chromium:702017, webrtc:7349
R=sprang@webrtc.org
Review-Url: https://codereview.webrtc.org/2860063002
Cr-Commit-Position: refs/heads/master@{#18017}
This change allows more callbacks to be registered to the test object.
The callbacks are used to give the user of the test object the ability
to instrument the test object. This CL specifically adds
instrumentation points just after a packet is inserted into NetEq, and
just after audio is pulled out of NetEq.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2851383004
Cr-Commit-Position: refs/heads/master@{#18014}
