When FixedGainController::SetGain() is called first on a large value (e.g., 40 dB)
and afterwards on a smaller one (e.g., 0 dB), the limiter used by FixedGainController
takes time (about 10-20 seconds) to converge. During that period, the audio is not
audible and the volume slowly increases.
Even if switching from 40 dB to 0 dB is unlikely, this behavior can be corrected by
resetting the limiter every time that FixedGainController::SetGain() is called.
This eliminates the undesired effect described above even when the transient is short.
Bug: webrtc:7494
Change-Id: I419b8986d2181448b4671cdbbd1c256dfb460216
Reviewed-on: https://webrtc-review.googlesource.com/94902
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24451}
Also adds a running picture id for the old generic format when
kVideoCodecGeneric is used (behind "WebRTC-GenericPictureId" field trial).
Bug: webrtc:9361
Change-Id: I6f232a2663bb60257c97ed3473eb07044d325b90
Reviewed-on: https://webrtc-review.googlesource.com/94842
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24449}
This CL adds adaptive handling of platforms where the echo path has
a strong gain above 10 kHz. A configurable offset is adaptively applied
depending on the amount of echo and mode of the echo suppressor.
Bug: webrtc:9663
Change-Id: I27dde6dc23b04a76a3be8c49d7fc9e226b9137e6
Reviewed-on: https://webrtc-review.googlesource.com/95947
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24448}
Disable the use of ERLE uncertainty with a diverged filter as it has
been shown to make transparency worse.
Bug: webrtc:9668
Change-Id: I5e23665def187c0d1cf47a029c4ebc950e79bb44
Reviewed-on: https://webrtc-review.googlesource.com/96140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24446}
Also consolidate lock operations to public methods only, moving one
CritScope out of UpdateCounters (private) up to IncomingPacket
(public).
Bug: webrtc:7135
Change-Id: I458857d3cfa49961fa34196ffe02cdefd83cec10
Reviewed-on: https://webrtc-review.googlesource.com/96122
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24443}
This CL allows selecting an echo suppressor behavior which is specific
for whether the nearend is dominant, or the echo is dominant.
The changes in this CL are bitexact.
Bug: webrtc:9660
Change-Id: Ie32e65efe47e692de6d6a22a7ad3b469d745fd6b
Reviewed-on: https://webrtc-review.googlesource.com/95725
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24434}
The AGC2 is enabled by flipping
AudioProcessing::Config::GainController2::enabled. The flag enables
both AdaptiveAgc and FixedGainController. Before this CL, there was no
way(*) to only enable the FixedGainController. After this CL, it's
also possible to flip the setting
|AudioProcessing::Config::GainController2::adaptive_digital_mode|. The
default is |true|, which is the previous behavior.
* Except for instantiating and setting it up outside of the APM like
it's done in the AudioMixer.
Bug: webrtc:7494
Change-Id: I506e93b6687221ac467f083fa8db3d45c98c1b83
Reviewed-on: https://webrtc-review.googlesource.com/95426
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24432}
This CL removes the UpdateGainIncrease code that is not used anymore.
The CL has been tested for bit exactness.
Bug: webrtc:8671
Change-Id: I4fcf26c3b4b5bba760ee139416ddefac86a36c2e
Reviewed-on: https://webrtc-review.googlesource.com/95940
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24425}
This CL removes some of the unused code in the suppressor. The CL has
been tested for bit exactness.
Bug: webrtc:8671
Change-Id: I960f9445dfd109cf1d5790debb8758872b5b8d0d
Reviewed-on: https://webrtc-review.googlesource.com/95682
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24417}
This CL adds a quiet mode for audioproc_f and hooks up the verbose
output of the AEC3 settings read from the JSON input file to that.
Bug: webrtc:8671
Change-Id: I93bbd1efc6502649da7b2b3e9f7557e9c184b0ed
Reviewed-on: https://webrtc-review.googlesource.com/95700
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24416}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I48b9deaad873aee597f56ebd33814420024e0d58
Reviewed-on: https://webrtc-review.googlesource.com/95645
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24405}
Videosendstream can be created before capturer starts, so initially the frame resolution may be zero. Add a check to prevent test failure and undesired behavior.
Bug: webrtc:7737
Change-Id: I8f4402e866f45ea1eb112437f866170691a111f6
Reviewed-on: https://webrtc-review.googlesource.com/95102
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24404}
This CL removes the unused coherence computation from AEC3. This CL
only removes unused code, the output of AEC3 does not change.
Bug: webrtc:8671
Change-Id: Ie127c5ec64e29414f1e1570511d57a4d09fc9145
Reviewed-on: https://webrtc-review.googlesource.com/95650
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24403}
This CL adds functionality for applying an optional fixed delay in AEC3
to the capture signal
Bug: webrtc:9647
Change-Id: Id3b3f896bcf203e6611298dc804c3c80da9f1883
Reviewed-on: https://webrtc-review.googlesource.com/95142
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24399}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I0e1462fa6e35944f7dbb02580f1db09401c8f7c8
Reviewed-on: https://webrtc-review.googlesource.com/95484
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24394}
This adds a start bitrate field to TargetRateConstraints since this is
required in VideoSendStreamTests. This also reduces the differences
between the old SendSideCongestionController and the new TaskQueue
based version.
Bug: webrtc:9586
Change-Id: I5d3d1414e9d30b51723c911a0bf2e96b876c04e5
Reviewed-on: https://webrtc-review.googlesource.com/92624
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24384}
This CL makes calls to send side congestion controller that originates
from the task queue execute directly rather than posting a task. This
ensures that side effects are applied by the time the call returns.
This reduces the risk that the task queue version of the congestion
controller introduces races that does not exist in the process thread
based version.
Bug: webrtc:9586
Change-Id: I82de032dc971c791a0f86d20ccbd47cbb09eba4b
Reviewed-on: https://webrtc-review.googlesource.com/85360
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24382}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.
I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.
Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}
This reverts commit 84916937b70472715efe5682bc273e91c3a72695.
Reason for revert: breaking downstream projects.
Original change's description:
> Update packetsLost and jitter stats any time a packet is received.
>
> Before this CL, the packetsLost and jitter stats (as returned by
> GetStats, at the API level) were only being updated when an RTCP SR or
> RR is generated. According to the stats spec, "local" stats like this
> should be updated any time a packet is received.
>
> This CL also fixes some minor issues with the calculation of packetsLost
> (and fractionLost):
> * Packets weren't being count as lost if lost over a sequence number
> rollover.
> * Temporary periods of "negative" loss (caused by duplicate or out of
> order packets) weren't being accumulated into the cumulative loss
> counter. Example:
> Period 1: Received packets 1, 2, 4
> Loss over that period: 1 (expected 4 packets, got 3)
> Reported cumulative loss: 1
> Period 2: Received packets 3, 5
> Loss over that period: -1 (expected 1 packet, got 2)
> Reported cumulative loss: 1 (should be 0!)
>
> Landing with NOTRY because Android compile bots are broken for an
> unrelated reason.
> NOTRY=True
>
> Bug: webrtc:8804
> Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
> Reviewed-on: https://webrtc-review.googlesource.com/50020
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23731}
TBR=danilchap@webrtc.org,deadbeef@webrtc.org,ossu@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Landing with NOTRY because ios64_sim_ios10_dbg bot is broken.
Passing all other bots.
NOTRY=True
Bug: webrtc:8804
Change-Id: I07bd6b1206d5a8d211792ad392842f9ed6c505e9
Reviewed-on: https://webrtc-review.googlesource.com/95280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24370}
The PCC congestion control algorithm divides time into consecutive
intervals called monitor intervals. This CL adds a class that is used by
PCC to measure the performance of sending at a certain rate during one
monitor interval.
Bug: webrtc:9434
Change-Id: Ia0447e224067d4ca807bcc6fd8083f9083385b91
Reviewed-on: https://webrtc-review.googlesource.com/87140
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24368}
This is a part of series of CLs adding PCC (Performance-oriented
Congestion Control).
Bug: webrtc:9434
Change-Id: Idd36d8abea008623ac64b939d0de7ee6001f7f23
Reviewed-on: https://webrtc-review.googlesource.com/86951
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24364}
This CL:
-Adds another config parameter that controls the duration of the initial
state.
-Adds reading of that parameter in audioproc_f from the json settings file.
-Adds missing reading of another parameter in audioproc_f from the json
settings file.
Bug: webrtc:8671
Change-Id: Ie6164c360492de5e6b0ade8838bbabe214560b5e
Reviewed-on: https://webrtc-review.googlesource.com/94621
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24360}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
Bug: none
Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
Reviewed-on: https://webrtc-review.googlesource.com/91861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24359}
This removes the reserved bitrate feature as it is not currently used.
This saves debugging time as there is less code to investigate. This
also makes it easier to compare the code with the task queue based
version which lacks this feature.
Bug: webrtc:9586
Change-Id: I207624ceb8d203c88c3d01bfc753d60523f99fe4
Reviewed-on: https://webrtc-review.googlesource.com/92622
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24357}
- Allow use of flexible mode which was blocked in webrtc:9261 since it
only worked together with old screen sharing. Since webrtc:9244 flexible mode
works with both normal and screen coding modes.
- Add unit test that checks that reference list encoder writes into RTP
payload descriptor and the predefined one match.
Bug: webrtc:9585
Change-Id: I4a1bdc51cbf15e7224cc7c271af8b2e3d46657d1
Reviewed-on: https://webrtc-review.googlesource.com/94778
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24355}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
It didn't account for implicit bitrate allocation, which is used in
some unit tests, when bitrate distribution is done by the encoder
wrapper.
Bug: none
Change-Id: I8fcf28e10f7a6c378580ef917221ad5c8d3869c9
Reviewed-on: https://webrtc-review.googlesource.com/94775
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24343}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I1be59a9277aad8f51765c26e34ab16b63bcaeb42
Reviewed-on: https://webrtc-review.googlesource.com/94774
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24340}
