This will be hooked up in clients who need to keep using the moderate
suppression level in AEC2 until other tuning options are available.
Bug: webrtc:9535
Change-Id: I6c40898954d9c856f58bcea87271f4b98fa124de
Reviewed-on: https://webrtc-review.googlesource.com/94148
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24292}
This CL introduces a few changes to the default VP8 temporal layers:
* The pattern is now reset on keyframes
* The sync flag is inferred rather than hard-coded
* Support is added for buffer search order
Bug: webrtc:9012
Change-Id: Ice19d32413d20982368a01a7d2540d155e185ad4
Reviewed-on: https://webrtc-review.googlesource.com/91863
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24288}
The FixedGainController is used in two places.
One is the AudioMixer. There it's used to limit the audio level after
adding streams. The other is GainController2, where it's placed after
steps that could boost the audio level outside the allowed range.
We log metrics from the FGC. To avoid confusion, this CL makes the two
use cases log to different histograms.
Chromium histogram CL is
https://chromium-review.googlesource.com/c/chromium/src/+/1170833
Bug: webrtc:7494
Change-Id: I1abe60fd8e96556f144d2ee576254b15beca1174
Reviewed-on: https://webrtc-review.googlesource.com/93464
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24284}
This CL changes the AEC3 code to allow the main and shadow filters
to have different lengths.
Bug: webrtc:9614,chromium:873100
Change-Id: I3ec2861d496986610d5a73db5771bbe9b8bf7dcd
Reviewed-on: https://webrtc-review.googlesource.com/93465
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24265}
This CL makes the jump-starting of the shadow filter more extreme.
It furthermore utilizes this to allow the AEC to rely further, and
more quickly on its linear filter estimates.
The result is mainly increased transparency but also some
cases of fewer echo blips.
Bug: webrtc:9612,chromium:873074
Change-Id: I90f7cfbff9acb9d0c36409593afbf476e7a830d3
Reviewed-on: https://webrtc-review.googlesource.com/93461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24264}
This CL also excludes several codec mappings depending on compile-time flags.
Bug: webrtc:9289
Change-Id: I1a9183f88378307925b747576a5513e54be3782e
Reviewed-on: https://webrtc-review.googlesource.com/93462
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24259}
This is a preparation for extracting CSRC book-keeping to its own
class.
Bug: webrtc:7135
Change-Id: Ic51ceb57ec53a43064a3d0392de8baa978a4e8cf
Reviewed-on: https://webrtc-review.googlesource.com/93463
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24257}
This CL ensures that the shadow filter is adapted at each block, which
avoids that a temporary filter length mismatch can occur between the
main and shadow filters.
Bug: webrtc:9602,chromium:872201
Change-Id: I651812b4e3b134c6c5e1fe3df5ab78dbdb5c1fb4
Reviewed-on: https://webrtc-review.googlesource.com/93000
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24253}
* Make shorter 4-frame pattern default if 2 temporal layers are used.
* Make DefaultTemporalLayers usable by upper simulcast stream with 2tl.
* If experimental settings are enable, bump the max bitrate for the top
stream. Since we're now using probing everywhere the rampup should be
less of an issue.
* Additionally, fixes an issue in full stack tests, where
ScopedFieldTrials in an experiment would override the
--force_fieldtrials specified at command line. Some trials added by
the test bots caused timeouts without this.
Bug: webrtc:9477
Change-Id: I42410605d416b51c4fbfe5b6b850997484af583c
Reviewed-on: https://webrtc-review.googlesource.com/92883
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24252}
Any modification of the render stream now happens *before* the
echo detector enqueues render stream frames. In this way, there
is no impact of the render pre-processor on the echo likelihood
metric.
Bug: webrtc:9591
Change-Id: I9b5e339e892796a0d0cd072fdd45d35ec89d8802
Reviewed-on: https://webrtc-review.googlesource.com/93031
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24251}
Also changes default value of frame ID in RTPVideoHeader to
kNoPictureId. Special care should be take so that picture ID will not
be set in RTPVideoHeader unless the client on the end supports
deserializing extended generic header.
Bug: webrtc:9582
Change-Id: Ib096373ed187f31e51d481193a2bda56de68f167
Reviewed-on: https://webrtc-review.googlesource.com/92084
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24250}
This CL adds a flag to optionally disable the digital gain control in
ExperimentalAgc. With the flag, Experimental Agc (henceforth AGC1)
only controls the adaptive analog gain. This flag can be combined to
that which activates AGC2. That way, one can enable the hybrid AGC
configuration AGC1 analog only + AGC2 fixed+adaptive digital.
Previously, there was a flag "use_agc2_digital_adaptive" in
AgcManagerDirect. Our ambition was that to activate the hybrid mode
described above with this flag. The behavior of the flag was not
implemented.
To activate the hybrid mode after this CL, set
ExperimentalAgc::digital_adaptive_disabled=true and
AudioProcessing::Config::GainController2::enabled=true.
We also add flags for these settings in audioproc_f.
Then the required settings are currently
audioproc_f --agc2 1 --agc 1 --experimental_agc 1 \
--experimental_agc_disable_digital_adaptive 1 \
-i [INPUT]
Bug: webrtc:7494
Change-Id: Iea798dc3899cec83d30ba71caba787262fcaef41
Reviewed-on: https://webrtc-review.googlesource.com/89740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24249}
The cumulative number of packets lost in a RTCP sender report can be
negative if there are duplicates. This CL fixes a bug that the parser of
RTCP reports treats the field as an unsigned integer, and incorrectly
reports large packet losses when a negative loss is reported.
Bug: webrtc:9601
Change-Id: I1109ac0741614d61bda743e13a390b7d3e147a9c
Reviewed-on: https://webrtc-review.googlesource.com/92942
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#24234}
Agc2 applies a digital gain to the nearend signal.
When the analog level changes, the digital gain calculation is no
longer valid. Therefore Agc2 should be notified to analog gain
changes.
This CL also allow audioproc_f to chain AGC1 and AGC2. In a dependent
CL we will allow using AGC1 for analog gain and AGC2 for digital
gain.
Bug: webrtc:7494
Change-Id: Id75b3728fbf2de1d84b7fba005e4670c7a2985d9
Reviewed-on: https://webrtc-review.googlesource.com/89387
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24231}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
To make it work, I had to add support for the "ptime" parameter to the
L16 codec.
Bug: webrtc:8396
Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126
Reviewed-on: https://webrtc-review.googlesource.com/87423
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24217}
* More canonical and efficient 'move to front'.
* Don't use 'new' when value semantic is fine.
* Simplify flow (remove One-off private method).
* Remove dead code.
Bug: webrtc:9575
Change-Id: Ie6a3c4e3d5e2342e77e54fd59fffa05f6e5f9ebe
Reviewed-on: https://webrtc-review.googlesource.com/92802
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24215}
This CL adds a factory for creating a GoogCC network controller that
can be used without RTCP specific messages. This prepares for enabling
use of other underlying protocols as long as they can provide per
packet feedback.
Bug: None
Change-Id: I6671181949d97abd18843d0f4edf75040cc3f007
Reviewed-on: https://webrtc-review.googlesource.com/84583
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24198}
This CL removes a check in RoundRobinPacketQueue::FinalizePop. This
check will trigger if a the pause state is changed in PacedSender while
a packet is sent. This is a rare occurrence but would yield flaky
behavior. The check should not be required for the code to function
since the paused state is not read in FinalizePop other than for this
check.
Bug: webrtc:9586
Change-Id: Ib9476168eb637dc2f9710d0592bed92c4b03dacb
Reviewed-on: https://webrtc-review.googlesource.com/92090
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24197}
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and
RtpPayloadRegistry.
Only intended change in behavior is that packets with unknown payload
types are not discarded at this level of the stack. They are discarded
higher up, in Channel::ReceivePacket (audio) and
RtpVideoStreamReceiver::ReceivePacket (video).
Bug: webrtc:8995
Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf
Reviewed-on: https://webrtc-review.googlesource.com/92087
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24196}
This reverts commit fad2aa23b406ca5d85b8aa9ab891f2067e51c782.
Reason for revert: There seems to be a mismatch with Chrome's default for VP8.
Original change's description:
> Extract color space from Vp8 decoder
>
> Makes use of ColorSpace class to extract info from Vp8 stream.
>
> Bug: webrtc:9522
> Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
> Reviewed-on: https://webrtc-review.googlesource.com/90183
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24086}
TBR=sprang@webrtc.org,emircan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9522
Change-Id: Ie589963159c9e7ccbc52bf3fdfcbc383656a4ca9
Reviewed-on: https://webrtc-review.googlesource.com/92500
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24191}
This changes makes the usage of the new probe controller reflect how the
old probe controller was used. That is probes are now sent as soon as
they are generated. This is to avoid regressions in performance doe to
the timing of the sent probes.
Bug: chromium:868776
Change-Id: I722585689258c9b01e8f1dc47249b284a05a2793
Reviewed-on: https://webrtc-review.googlesource.com/91441
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24175}
This CL adds functionality to use the shadow filter output instead
of the main filter output for cases when the former is better than
the latter. One case when that happens is when there have been an
echo path change, either in the acoustic path, in the audio buffers
or due to some active audio processing effects being applied on
the device.
The CL causes less echo leaks, in particular on devices with
active render processing.
Bug: webrtc:9581,chromium:869821
Change-Id: Icb8df1b94141598da82dc188051ac59e43338938
Reviewed-on: https://webrtc-review.googlesource.com/91820
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24166}
