13 Commits

Author SHA1 Message Date
Evan Shrubsole
d9dd939d66 Move safe_minmax.h to webrtc namespace
Bug: webrtc:42232595
Change-Id: Ia3d96dfe1b1c25b6cc21bbd99d24ded7461924cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378061
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43942}
2025-02-20 05:12:52 -08:00
Tommi
7f30dd11eb Remove deprecated methods
follow up to https://webrtc-review.googlesource.com/c/src/+/352582

Bug: chromium:335805780
Change-Id: I47f2842da9e86b686e3a3c2f4f28fa03d1cd297d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356241
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42589}
2024-07-04 13:19:15 +00:00
Tommi
51ad7c1277 Update FrameCombiner et al to use DeinterleavedView
* FrameCombiner is simpler. No additional channel pointers for buffers.
* Improve consistency in using views in downstream classes.
* Deprecate older methods (some have upstream dependencies).
* Use samples per channel instead of sample rate where the former is
  really what's needed.

Bug: chromium:335805780
Change-Id: I0dde8ed7a5a187bbddd18d3b6c649aa0865e6d4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352582
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42575}
2024-07-02 15:58:20 +00:00
Tommi
093824c4d2 Switch away from hz to samples per channel for FrameCombiner et al
This simplifies the following steps:
* FrameCombiner infers the sample rate from channel size
* Sends the inferred sample rate to FixedDigitalLevelEstimator
  and Limiter.
* Those classes then convert the sample rate to channel size.
  Along the way perform checks that the derived channel size value
  is a legal value (which has already been done by FrameCombiner).

To:
* FrameCombiner sends channel size to FixedDigitalLevelEstimator and
  Limiter.

Bug: chromium:335805780
Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42480}
2024-06-13 19:00:39 +00:00
Tommi
ff2bf4b195 Update FrameCombiner to use audio view methods for interleaved buffers
Along the way slightly simplify the class interface since views
carry audio properties. Also, now allocating FrameCombiner allocates
the mixing buffer in the same allocation.

Bug: chromium:335805780
Change-Id: Id7a76b040c11064e1e4daf01a371328769162554
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42465}
2024-06-12 09:44:40 +00:00
Ali Tofigh
f3592cb2a2 Adopt absl::string_view in modules/audio_processing/
Bug: webrtc:13579
Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37800}
2022-08-16 13:49:14 +00:00
Alessio Bazzica
5c3ae49b44 AudioFrameView: size_t -> int
Bug: webrtc:7494
Change-Id: I46b1328f3d7da721e144cc3752ed4f458084cf62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234522
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35163}
2021-10-07 14:41:03 +00:00
Alessio Bazzica
8aaa604375 AGC2 new data dumps
Bug: webrtc:7494
Change-Id: Id288dd426e1c2754805bc548fbffe0eaeaacf3da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213420
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33605}
2021-03-31 14:55:42 +00:00
Alessio Bazzica
76714a6cc8 AGC2 minor code clean up
Dead code removed plus const ref std::string to avoid copies.

Bug: webrtc:7494
Change-Id: Ic408a810ae310fea942f25fc697ab81017c8a739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201624
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32968}
2021-01-14 08:03:17 +00:00
Alessio Bazzica
3e4c77f1c1 Fix AGC2 fixed-adaptive gain controllers order.
This CL refactors AGC2 and fixes the order with which the fixed
and the adaptive digital gain controllers are applied - i.e., fixed
first, then adaptive and finally limiter.

FixedGainController has been removed since we need to split the
processing done by the gain applier and the limiter.
Also, GainApplier and Limiter are easy enough to be used without
a wrapper and a wrapper would need 2 separated calls in the right
order - i.e., error prone.

FrameCombiner in audio mixer has been adapted and now only uses the
limiter (which is what is needed since no gain is applied).

The unit tests for FixedGainController have been moved to
gain_controller2_unittests. They have been re-adapted and
ChangeFixedGainShouldBeFastAndTimeInvariant has been re-tuned.

Bug: webrtc:7494
Change-Id: I4d7daeae917257ac019a645b74deba6642f77322
Reviewed-on: https://webrtc-review.googlesource.com/c/108624
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25477}
2018-11-01 20:35:36 +00:00
Alessio Bazzica
746d46bec9 AGC2: renaming GainCurveApplier to Limiter.
Bug: webrtc:7494
Change-Id: I3dcfb864fd63dbf3f3e7345f8f4cac6c86987e8b
Reviewed-on: https://webrtc-review.googlesource.com/c/108581
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25436}
2018-10-30 16:00:18 +00:00
Alessio Bazzica
087e9bed41 AGC2 Limiter class renamed.
Limiter has been renamed to LimiterDbGainCurve, which is a more correct name
and will allow in a follow-up CL to reuse the Limiter name for GainCurveApplier.
This is done to allow to use the limiter without instancing the fixed digital
gain controller and then to fix an AGC2 issue (namely, fixed gain applied after
the adaptive one).

Bug: webrtc:7494
Change-Id: Icd7050e3e51b832bfbf35e5cc61109215c5b1ca6
Reviewed-on: https://webrtc-review.googlesource.com/c/106901
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25322}
2018-10-23 15:20:52 +00:00
Alex Loiko
a05ee82c4c Fixed Digital mode of AGC2 implementation finished.
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.

The other IGC and LE submodules were added in previous CLs [1] and
[2].

This CL also turns on AGC2 in the APM fuzzer.

[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381

Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:59:25 +00:00