The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.
A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.
BUG=webrtc:5609
Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
OnReceivedPacket now return the number of times the packet has been nacked. Also some minor refactoring.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972123002 .
Cr-Commit-Position: refs/heads/master@{#12717}
Reason for revert:
Breaks Chrome FYI using H264.
Need to investigate.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4170
Original issue's description:
> Remove ViEEncoder::SetNetworkStatus
>
> This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
>
> BUG=webrtc:5687
> NOTRY=True
>
> Committed: https://crrev.com/50b5c3be844ef571a28b2681c549443a26735d72
> Cr-Commit-Position: refs/heads/master@{#12699}
TBR=stefan@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/1978783002
Cr-Commit-Position: refs/heads/master@{#12715}
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.
Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
This CL will be followed with other CLs that break apart
the application of the comfort noise from the comfort
noise generation.
The changes in the CL are very close to bitexaxt. The
bitinexactness is caused by differences in numerical
behavior when bundling the spectral band power and the
noise scaling based on the NLP gain.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1958933002
Cr-Commit-Position: refs/heads/master@{#12713}
a separate method.
This CL will be followed by other CLs that simplify this method and break out the state specific to this computation
into a separate substate.
The changes are bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1963493003
Cr-Commit-Position: refs/heads/master@{#12712}
This CL implements the muted output functionality in NetEq. Tests are
added. The feature is currently off by default, and AcmReceiver makes
sure that the muted state is not engaged.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/1965733002
Cr-Commit-Position: refs/heads/master@{#12711}
Added conditional updating of the statistics and the delay estimate so that
updates are only done when the farend is non-stationary.
The reason for this is that all the values that go into the updating of the
statistics, and that in turn are also used to update the delay, are frozen
when the farend signal is non-stationary. Therefore, when the farend signal
is silent (stationary), the last estimates present before the silent (stationary)
period began are used to continue to update the statistics. This is a problem as
the updating is done in a manner that assumes that the estimates continue
to be updated.
This CL conditions the updating based on stationarity instead of silence
as both are treated in the same manner in the delay agnostic AEC.
This makes sense theoretically as the delay agnostic AEC operates on
analyzing power deviations (in bands) from a slowly updated average power and
therefore for a stationary signal will have no such deviations to base its analysis
on.
BUG=webrtc:5875, chromium:576624
NOTRY=True
Review-Url: https://codereview.webrtc.org/1967033002
Cr-Commit-Position: refs/heads/master@{#12700}
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
BUG=webrtc:5687
NOTRY=True
Review-Url: https://codereview.webrtc.org/1932683002
Cr-Commit-Position: refs/heads/master@{#12699}
Fixes bug where QualityScaler would be stuck "way below" QVGA (due to
downscale_shift_) even though it would never scale below QVGA. Also
fixes issue where samples would be cleared when either staying at max
resolution or going below QVGA even though no action happened.
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1971693003 .
Cr-Commit-Position: refs/heads/master@{#12691}
This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)
Original reverted cl is in patch set #1.
Changes in following patch sets.
The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()
It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True // Due to bug in android_x86 cq builder....
Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!
Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}
This is the last step in changing the signature of AudioEncoder::Encode
to taking an rtc::Buffer as its output parameter, rather than a pointer
to and a size parameter.
The notry parameter has been added specifically to work around android_compile_x86_dbg bot failing.
NOTRY=True
BUG=webrtc:5591
Review-Url: https://codereview.webrtc.org/1962013003
Cr-Commit-Position: refs/heads/master@{#12685}
Reason for revert:
Not perf-regression culprit.
Original issue's description:
> Revert of Rename OpenH264 frame-type conversion function. (patchset #2 id:20001 of https://codereview.webrtc.org/1943193003/ )
>
> Reason for revert:
> Speculative revert for perf regression (though unlikely).
>
> Original issue's description:
> > Rename OpenH264 frame-type conversion function.
> >
> > Also removing default case, so if another frame is added to
> > EVideoFrameType we have to handle it.
> >
> > This will now NOTREACHED on videoFrameTypeInvalid, but
> > videoFrameTypeInvalid shouldn't happen if encoding succeeds, so it
> > should be fine or we should become aware of it.
> >
> > BUG=
> > R=hbos@webrtc.org
> >
> > Committed: https://crrev.com/39a36705ab734914d500b8a0f214ea630d82ab70
> > Cr-Commit-Position: refs/heads/master@{#12636}
>
> TBR=hbos@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:610347
>
> Committed: https://crrev.com/1abf937cecea56ee02ac4a08980ffea9e7ed1054
> Cr-Commit-Position: refs/heads/master@{#12677}
TBR=hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:610347
Review-Url: https://codereview.webrtc.org/1970513004
Cr-Commit-Position: refs/heads/master@{#12679}
Reason for revert:
Speculative revert for perf regression (though unlikely).
Original issue's description:
> Rename OpenH264 frame-type conversion function.
>
> Also removing default case, so if another frame is added to
> EVideoFrameType we have to handle it.
>
> This will now NOTREACHED on videoFrameTypeInvalid, but
> videoFrameTypeInvalid shouldn't happen if encoding succeeds, so it
> should be fine or we should become aware of it.
>
> BUG=
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/39a36705ab734914d500b8a0f214ea630d82ab70
> Cr-Commit-Position: refs/heads/master@{#12636}
TBR=hbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:610347
Review URL: https://codereview.webrtc.org/1964913002 .
Cr-Commit-Position: refs/heads/master@{#12677}
AudioVector used NetEq was based on a shift buffer, which has a high complexity, and the complexity is very much dependent on the capacity of the buffer.
This CL changes the shift buffer to a ring buffer.
Reduction in the CPU usages of NetEq is expected.
BUG=608644
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1948483002 .
Cr-Commit-Position: refs/heads/master@{#12676}
Adding a new method to the Expand class, which will answer the question
whether an ongoing expansion has been faded down to zero
amplitude (i.e., been muted). Also adding a test.
This new functionality will be used in CLs to follow.
BUG=webrtc:5608
NOTRY=True
Review-Url: https://codereview.webrtc.org/1967473004
Cr-Commit-Position: refs/heads/master@{#12672}
This is to prepare for implementation of NetEq muted state, which may
cause GetAudioInternal to make an early return just before the call to
GetDecision. With this change, the stats are updated in any case.
BUG=webrtc:5608
NOTRY=True
Review-Url: https://codereview.webrtc.org/1948663002
Cr-Commit-Position: refs/heads/master@{#12671}
The timescale_holdoff_ is a counter in the DecisionLogic class. The
purpose is to enforce a minimum number of GetAudio calls
between (successfull) time-scaling operations (i.e., Accelerate and
Pre-emptive Expand operations). With this change, the counter is
replaced with a Countdown timer obtained from a TickTimer object.
BUG=webrtc:5608
R=tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1945863002 .
Cr-Commit-Position: refs/heads/master@{#12670}
because in the TMMBRHelp class it is independent of other members.
BUG=webrtc:5565
R=philipel
Review-Url: https://codereview.webrtc.org/1746773002
Cr-Commit-Position: refs/heads/master@{#12669}
The logging code was using the wrong constants for the
codec type, resulting in the type always being "unknown".
Tested: modules_unittests --gtest_filter='IvfFile*' -logs
BUG=
Review-Url: https://codereview.webrtc.org/1955273002
Cr-Commit-Position: refs/heads/master@{#12668}
files.
The purpose of this CL is to simplify upcoming AEC algorithm
changes.
The changes should be bitexact.
The presubmit was bypassed due to a presubmit complaint
about usage of short instead of int16_t which will be
addressed in upcoming CLs.
BUG=webrtc:5298, webrtc:5201
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/1949803004
Cr-Commit-Position: refs/heads/master@{#12662}
method in the AEC into two methods.
This CL is step towards simplifying the AEC code, making it
more modifiable and modular.
The changes should be bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1943753002
Cr-Commit-Position: refs/heads/master@{#12656}
method in the AEC.
This CL is step towards simplifying the AEC code, making it
more modifiable and modular.
The changes should be bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1943193002
Cr-Commit-Position: refs/heads/master@{#12655}
This CL is step towards simplifying the AEC code, making it more
modifiable and modular.
The changes should be bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1936203002
Cr-Commit-Position: refs/heads/master@{#12654}
theAEC. The reason for this is that this optimized method hinders any
refactoring of the code. In particular, it is not possible to separate
the application of the echo suppressor gain from the gain computation
and the comfort noise generation as all of these are partly included
in this method.
This CL is step towards simplifying the AEC code, making it more
modifiable and modular.
The changes should be bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1942853002
Cr-Commit-Position: refs/heads/master@{#12653}
This CL is step towards simplifying the AEC code, making it more modifiable and modular.
The changes should be bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1936173002
Cr-Commit-Position: refs/heads/master@{#12652}
There are still a few places in VideoReceiveStream where the RTP module
is explicitly used, e.g. setting up a/v sync, but it's a bigger task to
change and that will be done in a follow up instead of in this CL.
BUG=webrtc:5838
Review-Url: https://codereview.webrtc.org/1947913002
Cr-Commit-Position: refs/heads/master@{#12642}
the file content to echo_cancellation.h.
The purpose of this CL is to simplify upcoming AEC algorithm
changes.
The changes should be bitexact.
BUG=webrtc:5298, webrtc:5201
Review-Url: https://codereview.webrtc.org/1947743004
Cr-Commit-Position: refs/heads/master@{#12638}
Also removing default case, so if another frame is added to
EVideoFrameType we have to handle it.
This will now NOTREACHED on videoFrameTypeInvalid, but
videoFrameTypeInvalid shouldn't happen if encoding succeeds, so it
should be fine or we should become aware of it.
BUG=
R=hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/1943193003
Cr-Commit-Position: refs/heads/master@{#12636}
SSRC knowledge is contained withing VideoSendStream. That also means that debug recording is moved to VideoSendStream.
I think that make sence since that allows debug recording with external encoder implementations one day.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1936503002
Cr-Commit-Position: refs/heads/master@{#12632}
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca.
This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/
patchset #1 is a pure reland.
patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1947873002 .
Cr-Commit-Position: refs/heads/master@{#12630}
Reason for revert:
Previous mode aligns with other encoders, and RC_TIMESTAMP_MODE might have issues with no frames for several seconds.
Original issue's description:
> Use RC_TIMESTAMP_MODE for OpenH264.
>
> Performs rate control based on timestamp deltas instead of announced
> frame rate.
>
> BUG=webrtc:5855
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/c4deee49a3ec42b7fe83c82f750539b36aae1d3f
> Cr-Commit-Position: refs/heads/master@{#12611}
TBR=hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5855
Review-Url: https://codereview.webrtc.org/1950973002
Cr-Commit-Position: refs/heads/master@{#12629}
