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webrtc_m130/modules/audio_processing/agc/BUILD.gn

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Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../../webrtc.gni")
Remove AudioProcessing::gain_control() getter This change also resolves a bug in audioproc_f: The implicit ApplyConfig calls to enable gain control settings in aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig call on line 500 using a config from line 292. Compared to a ToT build including a fix for that bug, these changes are bitexact on a large number of aecdumps. [1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b Bug: webrtc:9878 Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 10:10:26 +02:00
rtc_source_set("gain_control_interface") {
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "gain_control.h" ]
Remove AudioProcessing::gain_control() getter This change also resolves a bug in audioproc_f: The implicit ApplyConfig calls to enable gain control settings in aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig call on line 500 using a config from line 292. Compared to a ToT build including a fix for that bug, these changes are bitexact on a large number of aecdumps. [1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b Bug: webrtc:9878 Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 10:10:26 +02:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("agc") {
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
sources = [
"agc_manager_direct.cc",
"agc_manager_direct.h",
]
configs += [ "..:apm_debug_dump" ]
deps = [
Remove AudioProcessing::gain_control() getter This change also resolves a bug in audioproc_f: The implicit ApplyConfig calls to enable gain control settings in aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig call on line 500 using a config from line 292. Compared to a ToT build including a fix for that bug, these changes are bitexact on a large number of aecdumps. [1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b Bug: webrtc:9878 Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 10:10:26 +02:00
":gain_control_interface",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
":level_estimation",
`AgcManagerDirect` ctor API and doc string improved Bug: chromium:1275566 Change-Id: Iedc8f5cbbf65fbf018da9df1aaa1f8ade1bbc063 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268840 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37569}
2022-07-19 12:18:38 +02:00
"..:api",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"..:apm_logging",
Making the Analog AGC properly support multi-channel This CL adds proper multi-channel support to the analog AGC. Beyond that, it prepares adding multi-channel support to the digital AGC by removing the tight dependency between the analog and digital AGC codes. Bug: webrtc:10859 Change-Id: I4414ccbc3db5dbb5ae069fdf426cbd038375ca7b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159480 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29878}
2019-11-22 12:11:40 +01:00
"..:audio_buffer",
Integrate ClippingPredictor into AudioProcessingImpl and AgcManagerDirect Integrate ClippingPredictor in AgcManagerDirect and AudioProcessingImpl. Disable functionality by default. Bug: webrtc:12774 Change-Id: Ic67a47f439c89b75066506fca8acaf636d8812f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221100 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34207}
2021-06-03 03:29:38 +02:00
"..:audio_frame_view",
Improve code quality in modules/audio_processing/agc/ - Switch from ptr+size to rtc::ArrayView - Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz - Stop passing nullptr in agc_manager_direct_unittest.cc when `AgcManagerDirect::Process()` is called - Allow to correctly run the tests added in the child CL (see [1]) [1] https://webrtc-review.googlesource.com/c/src/+/250141 Bug: webrtc:7494 Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35910}
2022-02-03 16:30:25 +01:00
"../../../api:array_view",
Removing all external access to the integer sample data in AudioBuffer This CL removes all external access to the integer sample data in the AudioBuffer class. It also removes the API in AudioBuffer that provides this. The purpose of this is to pave the way for removing the sample duplicating and implicit conversions between integer and floating point sample formats which is done inside the AudioBuffer. Bug: webrtc:10882 Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28912}
2019-08-20 09:19:21 +02:00
"../../../common_audio",
"../../../common_audio:common_audio_c",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"../../../rtc_base:checks",
Optionally disable digital gain control in ExperimentalAgc. This CL adds a flag to optionally disable the digital gain control in ExperimentalAgc. With the flag, Experimental Agc (henceforth AGC1) only controls the adaptive analog gain. This flag can be combined to that which activates AGC2. That way, one can enable the hybrid AGC configuration AGC1 analog only + AGC2 fixed+adaptive digital. Previously, there was a flag "use_agc2_digital_adaptive" in AgcManagerDirect. Our ambition was that to activate the hybrid mode described above with this flag. The behavior of the flag was not implemented. To activate the hybrid mode after this CL, set ExperimentalAgc::digital_adaptive_disabled=true and AudioProcessing::Config::GainController2::enabled=true. We also add flags for these settings in audioproc_f. Then the required settings are currently audioproc_f --agc2 1 --agc 1 --experimental_agc 1 \ --experimental_agc_disable_digital_adaptive 1 \ -i [INPUT] Bug: webrtc:7494 Change-Id: Iea798dc3899cec83d30ba71caba787262fcaef41 Reviewed-on: https://webrtc-review.googlesource.com/89740 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24249}
2018-08-09 15:04:24 +02:00
"../../../rtc_base:gtest_prod",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"../../../rtc_base:logging",
"../../../rtc_base:safe_minmax",
Adds WebRTC-Audio-AgcMinMicLevelExperiment to AGC1 Bug: webrtc:11065 Change-Id: Id07ebab7bfa12980187a5847d4f11c8a57450147 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158784 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29681}
2019-11-04 13:59:21 +01:00
"../../../system_wrappers:field_trial",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../../system_wrappers:metrics",
ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2 Bug: webrtc:7494 Change-Id: If88795fe34a73faa267a9c0bd5250e36455d4d81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277741 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38296}
2022-10-04 16:06:19 +02:00
"../agc2:clipping_predictor",
"../agc2:gain_map",
Fix WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget tests - Reset the tested metrics to avoid interactions between tests that depend on the execution order - Address the comment in [1] by adding a function to log the same histogram in two different places [1] https://chromium-review.googlesource.com/c/chromium/src/+/4087426/4/tools/metrics/histograms/metadata/web_rtc/histograms.xml#179 Bug: webrtc:7494 Change-Id: Ia4d339b03c8078eb63626c91579f8a9547f087f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287681 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38873}
2022-12-12 17:07:12 +01:00
"../agc2:input_volume_stats_reporter",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"../vad",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("level_estimation") {
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
sources = [
"agc.cc",
"agc.h",
"loudness_histogram.cc",
"loudness_histogram.h",
"utility.cc",
"utility.h",
]
deps = [
Improve code quality in modules/audio_processing/agc/ - Switch from ptr+size to rtc::ArrayView - Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz - Stop passing nullptr in agc_manager_direct_unittest.cc when `AgcManagerDirect::Process()` is called - Allow to correctly run the tests added in the child CL (see [1]) [1] https://webrtc-review.googlesource.com/c/src/+/250141 Bug: webrtc:7494 Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35910}
2022-02-03 16:30:25 +01:00
"../../../api:array_view",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"../../../rtc_base:checks",
"../vad",
]
}
APM: Change the AGC C code to be built as C++ code This CL changes so that the AGC legacy C code is built as C++. The CL also -removes #defines from the header files. -adds namespaces -removes unused code. To simplify the review, the CL is partitioned into different patchsets where each comprising of one step in the modification of the code (e.g., patch set 1 performs the renaming of the .c files to .cc). Bug: webrtc:5298 Change-Id: I362b17bde91142b2f2166acba4f2f888efd50fa1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171064 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30847}
2020-03-20 15:50:14 +01:00
rtc_library("legacy_agc") {
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
visibility = [
":*",
"..:*",
] # Only targets in this file and in
# audio_processing can depend on
# this.
sources = [
APM: Change the AGC C code to be built as C++ code This CL changes so that the AGC legacy C code is built as C++. The CL also -removes #defines from the header files. -adds namespaces -removes unused code. To simplify the review, the CL is partitioned into different patchsets where each comprising of one step in the modification of the code (e.g., patch set 1 performs the renaming of the .c files to .cc). Bug: webrtc:5298 Change-Id: I362b17bde91142b2f2166acba4f2f888efd50fa1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171064 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30847}
2020-03-20 15:50:14 +01:00
"legacy/analog_agc.cc",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"legacy/analog_agc.h",
APM: Change the AGC C code to be built as C++ code This CL changes so that the AGC legacy C code is built as C++. The CL also -removes #defines from the header files. -adds namespaces -removes unused code. To simplify the review, the CL is partitioned into different patchsets where each comprising of one step in the modification of the code (e.g., patch set 1 performs the renaming of the .c files to .cc). Bug: webrtc:5298 Change-Id: I362b17bde91142b2f2166acba4f2f888efd50fa1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171064 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30847}
2020-03-20 15:50:14 +01:00
"legacy/digital_agc.cc",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"legacy/digital_agc.h",
"legacy/gain_control.h",
]
deps = [
"../../../common_audio",
"../../../common_audio:common_audio_c",
Merge OOURA BUILD.gn files. No-Tree-Checks: True No-Try: True Bug: webrtc:11509 Change-Id: If66483a02085396e231986e39ef0dafcdd3f4bed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173740 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31114}
2020-04-21 09:56:16 +02:00
"../../../common_audio/third_party/ooura:fft_size_256",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"../../../rtc_base:checks",
Reland "Merge cpu_features build targets into //system_wrappers." This is a reland of 2b242d8fba59ccf67e4c5bcf5a1ea80917a23e1c Original change's description: > Merge cpu_features build targets into //system_wrappers. > > Before this CL, functions declared in cpu_features_wrapper.h where > not defined in the same build target, causing brittle builds that > might fail at link time if the binary was not depending on > //system_wrappers (the target with the definitions), violating [1]. > > This CL moves everything into //system_wrappers and also moves > cpu_features_wrapper.h definitions from C to C++ (in order to be able > to add the definitions to a C++ build target like //system_wrappers). > > [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs > > Bug: None > Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32039} TBR=kwiberg@webrtc.org Bug: None Change-Id: I1695b9a34d3ec20c50c1202a745f64fac58edef0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183444 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32042}
2020-09-05 21:55:35 +02:00
"../../../system_wrappers",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
]
if (rtc_build_with_neon) {
if (current_cpu != "arm64") {
Explicitly add -mfpu=neon to all targets that use NEON Remove obsolete comment about Chromium not defining NEON for Android. Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface". Bug: webrtc:9579 Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f Reviewed-on: https://webrtc-review.googlesource.com/91761 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 12:18:05 +02:00
# Enable compilation for the NEON instruction set.
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
}
}
if (rtc_include_tests) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("agc_unittests") {
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
testonly = true
sources = [
"agc_manager_direct_unittest.cc",
"loudness_histogram_unittest.cc",
"mock_agc.h",
]
configs += [ "..:apm_debug_dump" ]
deps = [
":agc",
Remove AudioProcessing::gain_control() getter This change also resolves a bug in audioproc_f: The implicit ApplyConfig calls to enable gain control settings in aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig call on line 500 using a config from line 292. Compared to a ToT build including a fix for that bug, these changes are bitexact on a large number of aecdumps. [1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b Bug: webrtc:9878 Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 10:10:26 +02:00
":gain_control_interface",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
":level_estimation",
"..:mocks",
Improve code quality in modules/audio_processing/agc/ - Switch from ptr+size to rtc::ArrayView - Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz - Stop passing nullptr in agc_manager_direct_unittest.cc when `AgcManagerDirect::Process()` is called - Allow to correctly run the tests added in the child CL (see [1]) [1] https://webrtc-review.googlesource.com/c/src/+/250141 Bug: webrtc:7494 Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35910}
2022-02-03 16:30:25 +01:00
"../../../api:array_view",
Add ClippingPredictor implementation Add implementation for clipping prediction and clipped level step estimation. Bug: webrtc:12774 Change-Id: I855d22980302aac7d49078ca29755f9422af9cb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220935 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34206}
2021-06-02 23:03:24 +02:00
"../../../rtc_base:checks",
Move random out of rtc_base_approved Bug: webrtc:9838 Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-14 12:41:26 +02:00
"../../../rtc_base:random",
AGC1 add clipping predictor evaluator Observes clipping predictions and detections and computes evaluation metrics for the predictor. Bug: webrtc:12774 Change-Id: I83f5942a3b6491de288510f2200f2f5c0e099bf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221619 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34305}
2021-06-16 14:55:40 +02:00
"../../../rtc_base:safe_conversions",
Disable Analog AGC based on the APM config Fixing a bug due to which the analog controller could not be disabled. AudioProcessing::Config::GainController1::AnalogGainController::enabled was ignored and therefore `recommended_stream_analog_level_locked()` in APM was returning the level recommended by `AgcManagerDirect`. When the analog controller is disabled, `stream_analog_level()` now returns the last value set via `set_stream_analog_level()`. However, the analog controller code is still running and, in particular, the existing metrics are reported as if the controller were enabled. This choice was made to reduce the risks of adding bugs in the digital compression gain selection part, which is tied to the analog controller. The metric drawback will be solved in a follow-up CL. Additional changes: - log `WebRTC.Audio.GainController.Analog.Enabled` when AGC1 is created or when its config changes - first step to replace "analog level" with "input volume" Bug: webrtc:7909, b/180019868 Change-Id: I28ce9556dd98f3dd9ad546799406c55478730435 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270663 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38044}
2022-09-05 16:04:31 +02:00
"../../../rtc_base:safe_minmax",
AGC1: min mic level override always applied When the minimum mic level is overridden via the field trial named WebRTC-Audio-AgcMinMicLevelExperiment, AGC1 can still lower the gain beyond the minimum value (namely, when clipping is observed). This CL changes the behavior of the field trial. When specified, the override always applies and therefore the mic level is guaranteed to never become lower than what the field trial specifies. Tested: RTC call in Chromium with and without --force-fieldtrials=" WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-255" Bug: chromium:1275566 Change-Id: I42ff45add54c11084f5ca6a2b95887c627c3c3aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250141 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35914}
2022-02-04 17:06:55 +01:00
"../../../rtc_base:stringutils",
Add AnalogGainStatsReporter to compute and report analog gain statistics Implement AnalogGainStatsReporter and add it in AudioProcessingImpl. This class computes statistics for analog gain updates and periodically reports them into a histogram. The added histograms for analog gain update statistics: - WebRTC.Audio.ApmAnalogGainDecreaseRate - WebRTC.Audio.ApmAnalogGainIncreaseRate - WebRTC.Audio.ApmAnalogGainUpdateRate - WebRTC.Audio.ApmAnalogGainDecreaseAverage - WebRTC.Audio.ApmAnalogGainIncreaseAverage - WebRTC.Audio.ApmAnalogGainUpdateAverage The histograms are defined in https://chromium-review.googlesource.com/c/chromium/src/+/3207987 Bug: webrtc:12774 Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35301}
2021-10-29 14:55:45 +02:00
"../../../system_wrappers:metrics",
Adds WebRTC-Audio-AgcMinMicLevelExperiment to AGC1 Bug: webrtc:11065 Change-Id: Id07ebab7bfa12980187a5847d4f11c8a57450147 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158784 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29681}
2019-11-04 13:59:21 +01:00
"../../../test:field_trial",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"../../../test:fileutils",
"../../../test:test_support",
"//testing/gtest",
]
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 14:44:38 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
}
}
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