Commit graph

17 commits

Author SHA1 Message Date
Gustaf Ullberg
d3ead1a942 AEC3: 'Block' class
This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code
easier to read and less error prone.

It also enables future changes to the underlying data structure of a
block. For instance, the data of all bands and channels could be stored
in a single vector.

The change has been verified to be bit-exact.

Bug: webrtc:14089
Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36968}
2022-05-23 09:53:46 +00:00
Byoungchan Lee
604fd2f1ab Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
2022-01-24 11:50:20 +00:00
Per Åhgren
6a05bb1b12 AEC3: Add signal dependent mixing before alignment
This CL adds code for doing signal-dependent downmixing
before the delay estimation in the multichannel case.

As part of the CL, the unittests of the render delay
controller are corrected. However, as that caused some of
them to fail, the CL (for now) as well disables the failing
test.

Bug: webrtc:11153,chromium:1029740, webrtc:11161
Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29980}
2019-12-03 11:30:48 +00:00
Gustaf Ullberg
ee84d39fce AEC3: Downmix multichannel signals before delay estimation
Multichannel signals are downmixed to mono before decimation and
delay estimation. This is useful when not all channels play
audio content. The feature can be toggled in the AEC3 configuration.

Bug: webrtc:10913
Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29126}
2019-09-10 08:16:07 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Gustaf Ullberg
777cf26328 AEC3: Clockdrift detection
This change introduces a clockdrift detector operating on the estimated
delay of the echo path delay estimator. Each time the delay estimate
changes it is compared to previous estimates. If the estimates are
slowly increasing or decreasing, clockdrift is detected.

Four different patterns are considered clockdrift:
- k, k+1, k+2, k+3
- k, k+2, k+1, k+3
- k, k-1, k-2, k-3
- k, k-2, k-1, k-3

A delay estimate history matching the three last elements in one of the
patterns is considered probable clockdrift. Matching all four elements
is considered verified clockdrift.

If the delay is constant for some time after clockdrift is detected the
clockdrift detector will revert to no detected clockdrift.

The level of clockdrift is reported via an UMA histogram.

Bug: webrtc:10014
Change-Id: I1cce4d593e101a8b3fa99df6935e59b4243cb97a
Reviewed-on: https://webrtc-review.googlesource.com/c/111381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25758}
2018-11-22 16:02:44 +00:00
Per Åhgren
8b7d206d37 AEC3: Decrease latency until the delay has been detected
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.

On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.

Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
2018-10-31 07:29:48 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Danil Chapovalov
db9f7ab9f9 Replace rtc::Optional with absl::optional in modules/audio processing
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_processing'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:38:56 +00:00
Per Åhgren
895ae9a0cd Improving the speed of the delay estimator in AEC3
This CL significantly improves the response time
of the AEC3 delay estimator to audio buffer issues.

The CL adds ensures that the delay estimator
correlators reacts to buffer issues from the
zero state which is much faster than if it has already
achieved a state matching a previous alignment.

The CL has been extensively tested on offline
recordings.

Bug: webrtc:9023, chromium:822245
Change-Id: Ic149b9429e592d4c3535eb8432582f435a1b4745
Reviewed-on: https://webrtc-review.googlesource.com/62081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22461}
2018-03-15 16:38:07 +00:00
Gustaf Ullberg
3646f973c2 AEC3 includes echo_canceller3_config.h directly
Avoid including audio_processing.h from within AEC3.

Bug: webrtc:8844
Change-Id: I02c475c2fb84e2c24eac86baac3c7edaa08bebc0
Reviewed-on: https://webrtc-review.googlesource.com/53065
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22029}
2018-02-15 08:30:14 +00:00
Per Åhgren
4712776bf4 Leveraging the skew in API call order to a boost AEC3 signal realignment
This CL resets the AEC3 realignment functionality when a significant
and persistent skew in the number of render and capture API calls is
detected.

Bug: chromium:811658,webrtc:8879
Change-Id: Ib5c727b38f427da2a7d25eac7c939a17bdaabe74
Reviewed-on: https://webrtc-review.googlesource.com/52260
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21997}
2018-02-13 12:52:58 +00:00
Per Åhgren
a76ef9d0b4 Robustify the faster alignment in AEC3 to avoid resets
The faster AEC3 alignment introduced recently may in
cases cause the alignment (and the AEC3) to repeatedly
reset. This CL avoids these resets by handling buffer
issues (which are triggering the resets) separately
during the initial coarse alignment phase.



Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53
Bug: webrtc:8798,chromium:805815
Reviewed-on: https://webrtc-review.googlesource.com/43480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21758}
2018-01-25 09:57:31 +00:00
Per Åhgren
38e2d95bda AEC3 delay estimator refactoring and introducing ability to customize
This CL refactors the delay estimator in AEC3.
Furthermore, it adds:
1. Allow for a customized delay estimator behavior to simplify
development.
2. Exposes that behavior to clear configuration settings.
3. Adds logging of the delay range supported by the delay
estimator.

Bug: webrtc:8519
Change-Id: I1764a090519a78b021b2e7de565c52a6c02c848e
Reviewed-on: https://webrtc-review.googlesource.com/21166
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20733}
2017-11-17 17:51:37 +00:00
Gustaf Ullberg
bd83b914c3 Separate AEC3 config from AudioProcessing::Config.
The struct containing the config for AEC3 is removed from
AudioProcessing::Config and is put in a new struct called
EchoCanceller3Config.

AEC3 should no longer be activated through
AudioProcessing::ApplyConfig. Instead an EchoCanceller3Factory
can be injected at AudioProcessing creation.

Bug: webrtc:8346
Change-Id: I27e3592e675eec3632a60c45d9e0d12514c2c567
Reviewed-on: https://webrtc-review.googlesource.com/11420
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20342}
2017-10-19 08:19:52 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h (Browse further)