Commit graph

8 commits

Author SHA1 Message Date
Alex Loiko
64cb83bbd9 Flags and settings for AGC2 in AgcManagerDirect.
This CL adds two flags to audioproc_f. The flags control
AgcManagerDirect. The flags are
'--experimental_agc_agc2_level_estimator' and
'--experimental_agc_agc2_digital_adaptive'.

After this CL, the flags are be applied to AgcManagerDirect. The flags
have no effect in release-mode. They cause a crash in debug-mode.

In an upcoming CL, '--experimental_agc_agc2_level_estimator 1' will
replace the speech level estimation in ExperimentalAgc with that of
AGC2.

'--experimental_agc_agc2_digital_adaptive 1' will replace the digital
gain selection and application with that of AGC2.

These audioproc_f will activate both new settings:

./out/Target/audioproc_f --agc 1 --experimental_agc 1
--experimental_agc_agc2_digital_adaptive 1
--experimental_agc_agc2_level_estimator 1 --simulate_mic_gain 1
--simulated_mic_kind 2

See also https://webrtc-review.googlesource.com/c/src/+/79360

Bug: webrtc:7494
Change-Id: If0e65893dffdddb312e553787b8cedaf9a334323
Reviewed-on: https://webrtc-review.googlesource.com/86548
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23802}
2018-07-02 13:20:39 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +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
Sam Zackrisson
af998e2fdc Remove non-API beamformer references
This removes beamformer references from audioproc_f, some non-beamformer tests, and a few other bits and bobs.
The beamformer is, after this, very decoupled from the remaining APM code.

Bug: webrtc:9402
Change-Id: Iaafc95517013d7a17723ef2329f17b5e09069bc9
Reviewed-on: https://webrtc-review.googlesource.com/83983
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23649}
2018-06-19 08:29:24 +00:00
Per Åhgren
c5efb0c080 Added an audioproc option to not report the stream delay
Bug: webrtc:9316
Change-Id: If7a20bbac998e9a779579650f3eb9019f974e9a8
Reviewed-on: https://webrtc-review.googlesource.com/79141
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23415}
2018-05-28 13:22:29 +00:00
Alex Loiko
5feb30e85f Options and settings for the Pre-amplifier.
Add configuration fields for the pre-amplifier in the Audio Processing
Module. Also add flags and settings for the pre-amplifier in
audioproc_f.

Also make the setting stored in Aec Dumps. And make the setting
applied when playing back Aec Dumps in audioproc_f.

Bug: webrtc:9138
Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3
Reviewed-on: https://webrtc-review.googlesource.com/69560
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22876}
2018-04-16 12:25:48 +00:00
Alex Luebs
24c220c178 Changed target_angle_degrees in audioproc_float to float to avoid integer division when converting to radians
Change-Id: I1b12d03524c34ed3fc4da89216539fd31a5c703b

Bug: none
Change-Id: I1b12d03524c34ed3fc4da89216539fd31a5c703b
Reviewed-on: https://webrtc-review.googlesource.com/61942
Commit-Queue: Alejandro Luebs <aluebs@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22462}
2018-03-15 19:01:47 +00:00
Ivo Creusen
2cb4105224 Moved audioproc_f interface into api directory.
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.

Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
2018-03-15 12:31:37 +00:00
Renamed from modules/audio_processing/test/audioproc_float.cc (Browse further)