This CL adds functionality to allow removal of any echo occurring
before the render and capture signals have been properly aligned.
The functionality is added in such a manner that the transparency
to nearend is maintained as much as possible.
Bug: webrtc:8883
Change-Id: I813cbbc4c48822e7dffcd9ab6233be4c222089de
Reviewed-on: https://webrtc-review.googlesource.com/49941
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22107}
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}
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}
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}
This CL changes the AEC3 behavior to be more transparent when there
is uncertainty about the amount of echo in the microphone signal.
Bug: webrtc:8398, chromium:774868
Change-Id: I88e681f8decd892f44397b753df371a1c4b90af0
Reviewed-on: https://webrtc-review.googlesource.com/10801
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20319}
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/render_delay_buffer.cc (Browse further)
