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}
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}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
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}
This CL adds functionality for passing the information about the
estimated delay to the echo remover in AEC3.
The CL also adds information about how long ago the delay changed,
and how long ago the delay estimate was updated.
Bug: webrtc:8671
Change-Id: If274ffe0465eb550f3e186d0599c6dc6fef7f5e8
Reviewed-on: https://webrtc-review.googlesource.com/55261
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22137}
This CL modifies the refactored render buffering scheme in AEC3
so that:
-A non-causal state can never occur which means that situations with
nonrecoverable echo should not occur.
-For a stable audio pipeline with a predefined API call jitter,
render overruns and underruns can never occur.
Bug: webrtc:8629,chromium:793305
Change-Id: I06ba1c368f92db95274090b08475dd02dbb85145
Reviewed-on: https://webrtc-review.googlesource.com/29861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21215}
This CL centralizes the render buffering in AEC3 so that all render
buffers are updated and synchronized/aligned with the render alignment
buffer.
Bug: webrtc:8597, chromium:790905
Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d
Reviewed-on: https://webrtc-review.googlesource.com/25680
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20989}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=henrik.lundin@webrtc.org
Bug: None
Change-Id: I733a83f702fe11884d229a1713cfac952727bde8
Reviewed-on: https://webrtc-review.googlesource.com/23601
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20786}
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}
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}
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}
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_remover_unittest.cc (Browse further)
