This CL introduces two related changes
1) It changes the way that the AEC3 determines whether the linear
filter is sufficiently good for its output to be used. The new scheme
achieves this much earlier than what was done in the legacy scheme.
2) It changes the way that saturated echo is and handled so that the
impact of the nearend speech is lower.
Bug: webrtc:9835,webrtc:9843,chromium:895435,chromium:895431
Change-Id: I0b493676886e2134205e9992bbe4badac7e414cc
Reviewed-on: https://webrtc-review.googlesource.com/c/104380
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25208}
This CL simplifies the buffering of render data. Instead of making assumptions
about the worst possible platform, it leverages recent improvements in
the delay estimator to quickly adapt when the conditions change.
Pros:
- No capture delay, delay is found ~200 ms faster.
- Cleaner code that makes the concept of delay more clear.
- Allows for removal of one matched filter because of the jitter headroom
removal.
Cons:
- Delay estimator needs to re-adapt when the call jitter increases.
The code can be deactivated by a kill switch. When the kill switch is
pulled the CL is bit exact.
Bug: webrtc:9726,chromium:895338
Change-Id: Ie2f9c8c5ce5b5a4510b4bdb95db2b970b57cd5d0
Reviewed-on: https://webrtc-review.googlesource.com/c/96920
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25169}
In this CL the use of the stationarity properties at init is set to true by default.
Bug: webrtc:9865, chromium:894439
Change-Id: I716ce0d792a50616dc38cc0ba6f2c702549a81cc
Reviewed-on: https://webrtc-review.googlesource.com/c/105303
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25123}
This CL extend critical sections to incorporate:
* private_submodules_->echo_controller
* config_
As a side benefit, it prevents weird interleaving where configuration
could have been changed in the middle of GetStatistics methods.
Bug: webrtc:9841
Change-Id: I0de5e756a684c2ff1be4effccf8c0f3d3175e3b9
Reviewed-on: https://webrtc-review.googlesource.com/c/105142
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25121}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This CL removes outdated code for testing of platforms with clock-drift
Bug: webrtc:8671
Change-Id: Ie202c514609d9f3d2357107b0daf895331275797
Reviewed-on: https://webrtc-review.googlesource.com/c/105183
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25105}
This CL ensures that the default setting for the usage of stationary signal
properties is not overridden by mistake.
Bug: chromium:894243
Change-Id: I85ab65383ee82b5f3153864da7a0cede7776c146
Reviewed-on: https://webrtc-review.googlesource.com/c/105181
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25104}
This CL changes the tuning of the echo suppressor for the case when
there is echo only. The resulting effect is a slight increase of
transparency
Bug: webrtc:9844,chromium:893744
Change-Id: I5e6a867e0d03dc3a468a8f5cfa64103e001baae1
Reviewed-on: https://webrtc-review.googlesource.com/c/104760
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25075}
This CL utilizes the AEC3 ability to tailor the suppressor during
situations when the nearend dominates over the residual echo. This is
done by increasing the thresholds for transparent echo suppressor
behavior when the nearend is strong compared to the residual echo.
Bug: webrtc:9836, chromium:893744
Change-Id: Ic06569eefc7f2557b401db43b3ac84b299071294
Reviewed-on: https://webrtc-review.googlesource.com/c/104460
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25071}
This is a new attempt to reduce the filter divergence
during double-talk without regressing in clock-drift
scenarios.
- The error_floor in decreased to allow for slow adaptation
when the filter performs well.
- The leakage_diverged is increased to allow for fast adaptation
when the shadow filter performs better.
- A new parameter, error_ceil, was added to stop the filter from
adapting too fast.
Bug: webrtc:9746,chromium:883264
Change-Id: Ie2868d2388b48412a192a004ec13f9eff34517b8
Reviewed-on: https://webrtc-review.googlesource.com/c/100460
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25063}
This CL lowers the default reverb decay to better match the standard
rooms where calls are made.
Bug: webrtc:9843
Change-Id: I46f1a629ecfdd72561829326d4fa58ede8107b6c
Reviewed-on: https://webrtc-review.googlesource.com/c/104740
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25061}
Original CL: https://webrtc-review.googlesource.com/c/src/+/101340
A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:
The histogram bins go from 0 to 100. But the value logged is dBFS. It
is always less than or equal to 0. This CL changes inverts the value
logged. The noise level value should be somewhere between -90 and 0
dBFS.
The histogram description is updated in
https://chromium-review.googlesource.com/c/chromium/src/+/1264578
Bug: webrtc:7494
Change-Id: I0b53630d4284ce1078fd453e05e89ee53ca9f6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/104063
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25021}
The AdaptiveAgc often boosts the signal outside of Float S16 range. It
is expected, which is why we have a limiter after it in the process
chain. But it turns out that this happens regularly even for simple
input examples. The output signal peaks can be as high as +4 dBFs for a
single speaker example (which should be easy). It leads to excessive
gain modulation by the limiter.
This CL is a new tuning designed to produce a safer gain. After this,
we shouldn't hit the saturation region of the limiter as often. But we
will still maintain a high gain.
We have a 'configurable kill-switch': the settings can be changed via
field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin.
Bug: webrtc:7494, chromium:892043
Change-Id: I5014377050c74c32ae8998282991141eae31cf58
Reviewed-on: https://webrtc-review.googlesource.com/c/102922
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25006}
This CL introduces a major refactoring of AecState for the purpose of
simplifying further improvements to the logic in this code.
The changes have successfully been tested for bitexactness.
Bug: webrtc:8671
Change-Id: If98efde55a22c76b093089a11a0562daac7e16e6
Reviewed-on: https://webrtc-review.googlesource.com/c/102362
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24996}
After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.
Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
This reverts commit 9e24dcff16.
Reason for revert: Breaks chromium.webrtc.fyi bots.
Original change's description:
> Export symbols needed by the Chromium component build (part 1).
>
> This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
> to mark WebRTC symbols as visible from a shared library, this doesn't
> mean these symbols are part of the public API (please continue to refer
> to [1] for info about what is considered public WebRTC API).
>
> [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
>
> Bug: webrtc:9419
> Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24969}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/103720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24974}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
Reviewed-on: https://webrtc-review.googlesource.com/c/103505
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24969}
This CL adds the ability to toggle the AGC2 adaptive digital mode in
audioproc_f
Bug: webrtc:5298
Change-Id: If1567d8c87f88992dff89253edb293a56cee0a73
Reviewed-on: https://webrtc-review.googlesource.com/c/103361
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24954}
This is a reland of 2fbb83b16b
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
Bug: webrtc:9535
Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/102881
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24943}
This CL corrects the incorrectly implemented check to avoid that AEC3
reacts on the initial pre-amp gain setting.
TBR: devicentepena@webrtc.org
Bug: webrtc:9805
Change-Id: I5decbf00a80457f24b8cd499c35720805ff9ccbc
Reviewed-on: https://webrtc-review.googlesource.com/c/103360
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24938}
In this CL the ERLE estimator is reset after a pre-amplifier gain change is communicated to APM.
Bug: webrtc:9805
Change-Id: I040f344e4607e862240250f9478d06de0d58a096
Reviewed-on: https://webrtc-review.googlesource.com/103222
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24933}
This change does three things:
- Move rtc_json into rtc_base/strings/, a non-API directory more fitting to
its purpose.
- Make a target for the currently unused json_unittest.
- Make the code available for use in non-test code again.
Bug: webrtc:9802
Change-Id: Id964a8a4b47b732a962a364894a4dbd3e7f4650f
Reviewed-on: https://webrtc-review.googlesource.com/103126
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24932}
Hypothetical scenario: short weak speech at start of call, then high
noise. The digital adaptive AGC2 would pick a high gain, and then
continue to apply it on the noise. Unless the noise is detected by the
noise estimator, the gain would never be reduced.
This CL addresses the issue by sending limiter gain info to the
adaptive digital AGC2.
Bug: webrtc:7494
Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64
Reviewed-on: https://webrtc-review.googlesource.com/102641
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24922}
This reverts commit 2fbb83b16b.
Reason for revert: Speculative revert over failing Chromium bot:
https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28M%20Nexus5X%29/117
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I1f8a27ac291f2cdc16c8daa32e399b74d489dbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/102642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24895}
These submodules implicitly rely on low cut filtering being enabled.
This CL clarifies a distinction:
High pass filtering is a feature that users can enable, according to the WebRTC standard.
Low cut filtering is a processing effect that is applied when any of the following is active:
- high pass filter
- noise suppression
- builtin echo cancellation
Bug: webrtc:9535
Change-Id: I9474276fb11354ea3b01e65a0699f6c29263770b
Reviewed-on: https://webrtc-review.googlesource.com/102600
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24892}
This is a sibling CL to a similar one for EchoCancellation:
https://webrtc-review.googlesource.com/c/src/+/97603
- EchoControlMobileImpl will no longer inherit EchoControlMobile.
- Removes usage of AudioProcessing::echo_control_mobile() inside most of
the audio processing module and unit tests.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (comfort noise, routing mode), but prints an
error message when unsupported settings are encountered.
Tested: audioproc_f with .wav and aecdump inputs.
Bug: webrtc:9535
Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
Reviewed-on: https://webrtc-review.googlesource.com/101621
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24888}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
This CL removes killswitches for code that has been properly tested in
experiments and is to be considered to be permanent.
The changes have been tested for bitexactness.
Bug: webrtc:8671
Change-Id: I0f9db16f377390d9dd3779096da91f3abc0fb4a5
Reviewed-on: https://webrtc-review.googlesource.com/102360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24877}
Specially for devices with high echo path gain, even low render signal can allow the linear filter of the AEC3 to converge. However, the conditions that were used for updating the ERLE avoided to update that estimation. In this commit, we allow adapting the ERLE estimator using even low render signal but the update of the ERLE is constraint in a way that decreases are not allowed.
Bug: webrtc:9776
Change-Id: Ic4331efcc47a0b05f394cdea9a88f336292de5a1
Reviewed-on: https://webrtc-review.googlesource.com/101641
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24859}
This reverts commit 3a9731ff2f.
Reason for revert: Seems to cause crashes in Chrome browser tests, see for example https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8934487169011818016/+/steps/browser_tests__retry_with_patch_/0/logs/WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsOfferEcdsaAnswerEcdsa/0
Original change's description:
> Bug in histogram metric reporting.
>
> A (actually several weeks) while ago, we noticed an error with the
> WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
> the value 0. Here is why:
>
> The histogram bins go from 0 to 100. But the value logged is dBFS. It is
> always less than or equal to 0. This CL changes the bins.
>
> Bug: webrtc:7494
> Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
> Reviewed-on: https://webrtc-review.googlesource.com/101340
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24800}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I84883f73710b7e13aa90ee29b140acfc417f109f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/101701
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24809}
Speeds up adaptation of the matched filter of the delay estimator by
allowing the estimated echo and the error signal (microphone minus
estimated echo) to be saturated. Only microphone saturation pauses
the filter adaptation.
Bug: webrtc:9773
Change-Id: I8b8400539fde3ee821f36a95818bece02ddd626b
Reviewed-on: https://webrtc-review.googlesource.com/101341
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24802}
A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:
The histogram bins go from 0 to 100. But the value logged is dBFS. It is
always less than or equal to 0. This CL changes the bins.
Bug: webrtc:7494
Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
Reviewed-on: https://webrtc-review.googlesource.com/101340
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24800}
Right after a volume decrease, the echo path estimate is overestimated and, as a side effect, the nearend signal is also overestimated. Due to that, the suppression gains are kept high avoiding the suppression of echoes. In this CL the neared power spectrum estimation is limited to a level given by the power spectrum or the microphone input signal. Additionally, the minimum gain that is computed inside the suppressor is also modified. Instead of using the nearend power spectrum that is now bounded, the power spectrum of the signal after the linear echo canceler is used.
Bug: webrtc:9762
Change-Id: Ia24cd2ce248f2c2ba124711b75acff3b8c5cfa9f
Reviewed-on: https://webrtc-review.googlesource.com/100720
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24796}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
Original CL:
https://webrtc-review.googlesource.com/c/src/+/97603
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Revert CL:
https://webrtc-review.googlesource.com/c/src/+/100305
Bug: webrtc:9535
TBR: gustaf@webrtc.org
Change-Id: I9248046b3a6a82df6221e502481836948643a991
Reviewed-on: https://webrtc-review.googlesource.com/100461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24749}
This CL:
-Adds the option to log the aec3 parameters used for a simulation.
-Cleans up the logging of the custom setting of aec3 parameters to
instead rely on the newly added logging.
Bug: webrtc:8671
Change-Id: If73a73d08e5a5077416033ded598a83fb1ade3e0
Reviewed-on: https://webrtc-review.googlesource.com/100381
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24742}
The test is refitted to use the AudioProcessingStats struct to get
reference data.
The old metrics do not map entirely injectively to the new ones, so the
reference protobuf and files are updated as well.
Bug: webrtc:9535
Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36
Reviewed-on: https://webrtc-review.googlesource.com/100100
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24740}
This reverts commit 1a03960e63.
Reason for revert: breaks downstream projects.
Original change's description:
> Remove APM internal usage of EchoCancellation
>
> This CL:
> - Changes EchoCancellationImpl to inherit privately from
> EchoCancellation.
> - Removes usage of AudioProcessing::echo_cancellation() inside most of
> the audio processing module and unit tests.
> - Default-enables metrics collection in AEC2.
>
> This CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (drift compensation, suppression level), but
> prints an error message when such settings are encountered.
>
> Some code in audio_processing_unittest.cc still uses the old interface.
> I'll handle that in a separate change, as it is not as straightforward
> to preserve coverage.
>
> Bug: webrtc:9535
> Change-Id: Ia4d4b8d117ccbe516e5345c15d37298418590686
> Reviewed-on: https://webrtc-review.googlesource.com/97603
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24724}
TBR=gustaf@webrtc.org,saza@webrtc.org
Change-Id: Ifdc4235f9c5ee8a8a5d32cc8e1dda0853b941693
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/100305
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24729}
This CL:
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
This CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Some code in audio_processing_unittest.cc still uses the old interface.
I'll handle that in a separate change, as it is not as straightforward
to preserve coverage.
Bug: webrtc:9535
Change-Id: Ia4d4b8d117ccbe516e5345c15d37298418590686
Reviewed-on: https://webrtc-review.googlesource.com/97603
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24724}
