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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 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.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
Import proto_library.gni when rtc_enable_protobuf is true Import proto_library.gni when rtc_enable_protobuf is true instead of when build_with_mozilla is false. Makes it maybe easier to reason about the intention (e.g. intention is to not compile any protobuf in, hence flag rtc_enable_protobuf) The build file could not work if build_with_mozilla = true but rtc_enable_protobuf = true. Bug: webrtc:10338 Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597 Reviewed-on: https://webrtc-review.googlesource.com/c/123763 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26868}
2019-02-26 11:46:17 +02:00
if (rtc_enable_protobuf) {
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
import("//third_party/protobuf/proto_library.gni")
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
config("apm_debug_dump") {
if (apm_debug_dump) {
defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
} else {
defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
}
}
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("config") {
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
visibility = [ ":*" ]
sources = [
"include/config.cc",
"include/config.h",
]
deps = [
"../../rtc_base:macromagic",
"../../rtc_base/system:rtc_export",
]
}
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("api") {
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
visibility = [ "*" ]
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
sources = [
"include/audio_processing.cc",
"include/audio_processing.h",
]
deps = [
":audio_frame_view",
":audio_generator_interface",
":audio_processing_statistics",
":config",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../../api/audio:aec3_config",
APM: Localize/abstract the usage of AudioFrame This CL moves the implementation of of the AudioFrame support from the implementation of AudioProcessing to proxy methods that map the call to the integer stream interfaces (added in another CL). The CL also changes the WebRTC code using the AudioFrame interfaces to instead use the proxy methods. This CL will be followed by one more CL that removes the usage of the AudioFrame class from the rest of APM (apart from the AudioProcessing API). Bug: webrtc:5298 Change-Id: Iecb72e9fa896ebea3ac30e558489c1bac88f5891 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170110 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30812}
2020-03-17 13:23:58 +01:00
"../../api/audio:audio_frame_api",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../../api/audio:echo_control",
"../../rtc_base:deprecation",
"../../rtc_base:macromagic",
"../../rtc_base:rtc_base_approved",
Adding more refined control over choice of band-splitting This CL allows the user to have more refined control over what band splitting-scheme is used inside the audio processing module. Bug: webrtc:6181 Change-Id: I236c3b1f96ab80cc4ffb8c39c045c034764567a1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152480 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29189}
2019-09-15 00:27:58 +02:00
"../../rtc_base/system:arch",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../../rtc_base/system:rtc_export",
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
"agc:gain_control_interface",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"//third_party/abseil-cpp/absl/types:optional",
]
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
}
APM: Localize/abstract the usage of AudioFrame This CL moves the implementation of of the AudioFrame support from the implementation of AudioProcessing to proxy methods that map the call to the integer stream interfaces (added in another CL). The CL also changes the WebRTC code using the AudioFrame interfaces to instead use the proxy methods. This CL will be followed by one more CL that removes the usage of the AudioFrame class from the rest of APM (apart from the AudioProcessing API). Bug: webrtc:5298 Change-Id: Iecb72e9fa896ebea3ac30e558489c1bac88f5891 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170110 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30812}
2020-03-17 13:23:58 +01:00
rtc_library("audio_frame_proxies") {
visibility = [ "*" ]
sources = [
"include/audio_frame_proxies.cc",
"include/audio_frame_proxies.h",
]
deps = [
":api",
":audio_frame_view",
"../../api/audio:audio_frame_api",
]
}
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("audio_buffer") {
Adding a placeholder audio_buffer build target This CL adds a placeholder build target in preparation for an upcoming CL (https://webrtc-review.googlesource.com/c/src/+/125081). Bug: webrtc:10366 Change-Id: I5b226e01d561689acf1624e2c0bad30cc1865011 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125560 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26965}
2019-03-05 08:40:48 +01:00
visibility = [ "*" ]
Make AEC3 the default desktop AEC option in WebRTC Bug: webrtc:10366 Change-Id: I854ed62df1da489fdab43e9157dff79b7287cacb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125081 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26983}
2019-03-06 04:16:46 +01:00
configs += [ ":apm_debug_dump" ]
Adding a placeholder audio_buffer build target This CL adds a placeholder build target in preparation for an upcoming CL (https://webrtc-review.googlesource.com/c/src/+/125081). Bug: webrtc:10366 Change-Id: I5b226e01d561689acf1624e2c0bad30cc1865011 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125560 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26965}
2019-03-05 08:40:48 +01:00
sources = [
Make AEC3 the default desktop AEC option in WebRTC Bug: webrtc:10366 Change-Id: I854ed62df1da489fdab43e9157dff79b7287cacb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125081 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26983}
2019-03-06 04:16:46 +01:00
"audio_buffer.cc",
"audio_buffer.h",
"splitting_filter.cc",
"splitting_filter.h",
"three_band_filter_bank.cc",
"three_band_filter_bank.h",
]
defines = []
deps = [
":api",
Optimizations and refactoring of the APM 3-band split filter This CL refactors and optimizes the 3-band split-filter in APM, which is a very computationally complex component. Beyond optimizing the code, the filter coefficients are also quantized to avoid denormals. The changes reduces the complexity of the split filter by about 30-50%. The CL has been tested for bitexactness on a number of aecdump recordings. (the CL also removes the now unused code for the sparse_fir_filter) Bug: webrtc:6181 Change-Id: If45f8d1f189c6812ccb03721156c77eb68181211 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168189 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30592}
2020-02-21 13:31:07 +01:00
"../../api:array_view",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Make AEC3 the default desktop AEC option in WebRTC Bug: webrtc:10366 Change-Id: I854ed62df1da489fdab43e9157dff79b7287cacb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125081 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26983}
2019-03-06 04:16:46 +01:00
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
Adding a placeholder audio_buffer build target This CL adds a placeholder build target in preparation for an upcoming CL (https://webrtc-review.googlesource.com/c/src/+/125081). Bug: webrtc:10366 Change-Id: I5b226e01d561689acf1624e2c0bad30cc1865011 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125560 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26965}
2019-03-05 08:40:48 +01:00
]
}
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("high_pass_filter") {
Use the AEC3 high-pass filter for the whole APM This CL removes and replaces the legacy fixed-point high-pass filter in APM with the floating point high-pass filter in AEC3. Bug: webrtc:10907 Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28950}
2019-08-23 21:29:17 +02:00
visibility = [ "*" ]
sources = [
"high_pass_filter.cc",
"high_pass_filter.h",
]
defines = []
deps = [
":audio_buffer",
"../../api:array_view",
"../../rtc_base:checks",
"utility:cascaded_biquad_filter",
]
}
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("audio_processing") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
configs += [ ":apm_debug_dump" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
"audio_processing_impl.cc",
"audio_processing_impl.h",
"common.h",
"echo_control_mobile_impl.cc",
"echo_control_mobile_impl.h",
Add algorithm for Residual Echo Detector. This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 07:04:03 -07:00
"echo_detector/circular_buffer.cc",
"echo_detector/circular_buffer.h",
"echo_detector/mean_variance_estimator.cc",
"echo_detector/mean_variance_estimator.h",
Added a new echo likelihood stat that reports the maximum value from a previous time period. BUG=webrtc:6797 Review-Url: https://codereview.webrtc.org/2629563003 Cr-Commit-Position: refs/heads/master@{#16079}
2017-01-15 08:29:46 -08:00
"echo_detector/moving_max.cc",
"echo_detector/moving_max.h",
Add algorithm for Residual Echo Detector. This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 07:04:03 -07:00
"echo_detector/normalized_covariance_estimator.cc",
"echo_detector/normalized_covariance_estimator.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"gain_control_impl.cc",
"gain_control_impl.h",
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
"gain_controller2.cc",
"gain_controller2.h",
Merge :audio_processing and :aec_dump_interface. Merges the two targets in modules/audio_processing and removes some redundant code. This enables not writing a bunch of redundant code in https://webrtc-review.googlesource.com/c/src/+/70502 ':audio_processing' did depend on ':aec_dump_interface'. 'modules/audio_processing/aec_dump' did depend on 'aec_dump_interface' but not ':audio_processing'. Having the AecDump implementation not depending on 'audio_processing' allows to have faster compilation time and reduces the dependencies. However, maintaining such a decoupling makes APM and AecDump client code more complex. NOTRY=true # want this in and 'ios_api_framework' seems stuck. Bug: webrtc:7404 Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c Reviewed-on: https://webrtc-review.googlesource.com/76603 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23244}
2018-05-15 13:13:22 +02:00
"include/aec_dump.cc",
"include/aec_dump.h",
Remove AudioProcessing::level_estimator() getter The new configuration path is via AudioProcessing::ApplyConfig and AudioProcessing::GetStatistics. Bug: webrtc:9878 Change-Id: Ic912d67455fcef4895566edb8fef62baf62d7cfe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156440 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29454}
2019-10-11 19:31:07 +02:00
"level_estimator.cc",
"level_estimator.h",
Removed the ProcessingComponent class BUG= Review URL: https://codereview.webrtc.org/1772553002 Cr-Commit-Position: refs/heads/master@{#11950}
2016-03-10 23:05:28 -08:00
"render_queue_item_verifier.h",
Add empty residual echo detector. This CL does not contain the actual algorithm, but only creates an empty processing component and connects the right signals to it. The algorithm will be added in a follow-up CL. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2405403003 Cr-Commit-Position: refs/heads/master@{#14820}
2016-10-28 05:39:16 -07:00
"residual_echo_detector.cc",
"residual_echo_detector.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"typing_detection.cc",
"typing_detection.h",
]
defines = []
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
":api",
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
":apm_logging",
Adding a placeholder audio_buffer build target This CL adds a placeholder build target in preparation for an upcoming CL (https://webrtc-review.googlesource.com/c/src/+/125081). Bug: webrtc:10366 Change-Id: I5b226e01d561689acf1624e2c0bad30cc1865011 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125560 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26965}
2019-03-05 08:40:48 +01:00
":audio_buffer",
APM: Localize/abstract the usage of AudioFrame This CL moves the implementation of of the AudioFrame support from the implementation of AudioProcessing to proxy methods that map the call to the integer stream interfaces (added in another CL). The CL also changes the WebRTC code using the AudioFrame interfaces to instead use the proxy methods. This CL will be followed by one more CL that removes the usage of the AudioFrame class from the rest of APM (apart from the AudioProcessing API). Bug: webrtc:5298 Change-Id: Iecb72e9fa896ebea3ac30e558489c1bac88f5891 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170110 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30812}
2020-03-17 13:23:58 +01:00
":audio_frame_proxies",
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
":audio_frame_view",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
":audio_generator_interface",
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
":audio_processing_statistics",
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
":config",
Use the AEC3 high-pass filter for the whole APM This CL removes and replaces the legacy fixed-point high-pass filter in APM with the floating point high-pass filter in AEC3. Bug: webrtc:10907 Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28950}
2019-08-23 21:29:17 +02:00
":high_pass_filter",
Add empty placeholder target for rms_level.* A follow-up CL will move the rms_level.* files into the new target. Bug: webrtc:11226 Change-Id: I59579b026346f627c0a2739d25f90c12bffbf248 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171102 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30840}
2020-03-20 11:30:32 +01:00
":rms_level",
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
":voice_detection",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../../api:function_view",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:aec3_config",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:echo_control",
Move functionality out from AudioFrame and into AudioFrameOperations. This CL is in preparation to move the AudioFrame into webrtc/api. The AudioFrame is a POD type used for representing 10ms of audio. It appears as a parameter and return value of interfaces being migrated to webrtc/api, in particular AudioMixer. Here, methods operator+=, operator>>=, Mute are moved into a new target webrtc/audio/utility/audio_frame_operations, and dependencies are changed to use the new versions. The old AudioFrame methods are marked deprecated. The audio frame utilities in webrtc/modules/utility:audio_frame_operations are also moved to the new location. TBR=kjellander@webrtc.org BUG=webrtc:6548 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2424173003 Cr-Commit-Position: refs/heads/master@{#15413}
2016-12-05 01:46:09 -08:00
"../../audio/utility:audio_frame_operations",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio/third_party/fft4g",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:deprecation",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Reland "Export symbols needed by the Chromium component build (part 1)." This reverts commit 16fe3f290a524a136f71660a114d0b03ef501f10. Reason for revert: After discussing this problem with nisse@ and yvesg@, we decided to modify how RTC_EXPORT works and avoid to depend on the macro COMPONENT_BUILD. RTC_EXPORT will instead depend on a macro WEBRTC_COMPONENT_BUILD (which can be set as a GN argument which defaults to false). When all the symbols needed by Chromium will be marked with RTC_EXPORT we will flip the GN arg in Chromium, setting to to `component_build` and from that moment, Chromium will depend on a WebRTC shared library when `component_build=true`. Original change's description: > Revert "Export symbols needed by the Chromium component build (part 1)." > > This reverts commit 99eea42fc1fe0be0ebed13c5eba7e1e42059bc5a. > > Reason for revert: > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::UnwrapTurnPacket(unsigned char const *, unsigned int, unsigned int *, unsigned int *)" (__imp_?UnwrapTurnPacket@cricket@@YA_NPBEIPAI1@Z) > >>> referenced by obj/services/network/network_service/socket_manager.obj:("virtual void __thiscall network::P2PSocketManager::DumpPacket(class base::span<unsigned char const, 4294967295>, bool)" (?DumpPacket@P2PSocketManager@network@@EAEXV?$span@$$CBE$0PPPPPPPP@@base@@_N@Z)) > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ValidateRtpHeader(unsigned char const *, unsigned int, unsigned int *)" (__imp_?ValidateRtpHeader@cricket@@YA_NPBEIPAI@Z) > >>> referenced by obj/services/network/network_service/socket_manager.obj:("virtual void __thiscall network::P2PSocketManager::DumpPacket(class base::span<unsigned char const, 4294967295>, bool)" (?DumpPacket@P2PSocketManager@network@@EAEXV?$span@$$CBE$0PPPPPPPP@@base@@_N@Z)) > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ApplyPacketOptions(unsigned char *, unsigned int, struct rtc::PacketTimeUpdateParams const &, unsigned __int64)" (__imp_?ApplyPacketOptions@cricket@@YA_NPAEIABUPacketTimeUpdateParams@rtc@@_K@Z) > >>> referenced by obj/services/network/network_service/socket_tcp.obj:("virtual void __thiscall network::P2PSocketTcp::DoSend(class net::IPEndPoint const &, class std::vector<signed char, class std::allocator<signed char>> const &, struct rtc::PacketOptions const &, struct net::NetworkTrafficAnnotationTag)" (?DoSend@P2PSocketTcp@network@@MAEXABVIPEndPoint@net@@ABV?$vector@CV?$allocator@C@std@@@std@@ABUPacketOptions@rtc@@UNetworkTrafficAnnotationTag@4@@Z)) > >>> referenced by obj/services/network/network_service/socket_tcp.obj:("virtual void __thiscall network::P2PSocketStunTcp::DoSend(class net::IPEndPoint const &, class std::vector<signed char, class std::allocator<signed char>> const &, struct rtc::PacketOptions const &, struct net::NetworkTrafficAnnotationTag)" (?DoSend@P2PSocketStunTcp@network@@MAEXABVIPEndPoint@net@@ABV?$vector@CV?$allocator@C@std@@@std@@ABUPacketOptions@rtc@@UNetworkTrafficAnnotationTag@4@@Z)) > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ApplyPacketOptions(unsigned char *, unsigned int, struct rtc::PacketTimeUpdateParams const &, unsigned __int64)" (__imp_?ApplyPacketOptions@cricket@@YA_NPAEIABUPacketTimeUpdateParams@rtc@@_K@Z) > >>> referenced by obj/services/network/network_service/socket_udp.obj:("bool __thiscall network::P2PSocketUdp::DoSend(struct network::P2PSocketUdp::PendingPacket const &)" (?DoSend@P2PSocketUdp@network@@AAE_NABUPendingPacket@12@@Z)) > > Original change's description: > > Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" > > > > This reverts commit b49520bfc08f5c5832dda1d642125f0bb898f974. > > > > Reason for revert: Problem fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1261398. > > > > Original change's description: > > > Revert "Reland "Export symbols needed by the Chromium component build (part 1)."" > > > > > > This reverts commit 588f4642d1a29f7beaf28265dbd08728191b4c52. > > > > > > Reason for revert: Breaks WebRTC Chromium FYI Win Builder (dbg). > > > lld-link: error: undefined symbol: "__declspec(dllimport) __thiscall webrtc::Config::Config(void)" (__imp_??0Config@webrtc@@QAE@XZ) > > > [...] > > > > > > Original change's description: > > > > Reland "Export symbols needed by the Chromium component build (part 1)." > > > > > > > > This reverts commit 2ea9af227517556136fd629dd2663c0d75d77c7b. > > > > > > > > Reason for revert: The problem will be fixed by > > > > https://chromium-review.googlesource.com/c/chromium/src/+/1261122. > > > > > > > > Original change's description: > > > > > Revert "Export symbols needed by the Chromium component build (part 1)." > > > > > > > > > > This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53. > > > > > > > > > > Reason for revert: Breaks chromium.webrtc.fyi bots. > > > > > > > > > > Original change's description: > > > > > > Export symbols needed by the Chromium component build (part 1). > > > > > > > > > > > > This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) > > > > > > to mark WebRTC symbols as visible from a shared library, this doesn't > > > > > > mean these symbols are part of the public API (please continue to refer > > > > > > to [1] for info about what is considered public WebRTC API). > > > > > > > > > > > > [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md > > > > > > > > > > > > Bug: webrtc:9419 > > > > > > Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99 > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103505 > > > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > > > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > > > > > Cr-Commit-Position: refs/heads/master@{#24969} > > > > > > > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > > > > > > > Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f > > > > > No-Presubmit: true > > > > > No-Tree-Checks: true > > > > > No-Try: true > > > > > Bug: webrtc:9419 > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103720 > > > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#24974} > > > > > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > > > > > Change-Id: I83bbc7f550fc23e823c4d055e0a6f60c828960dd > > > > No-Presubmit: true > > > > No-Tree-Checks: true > > > > No-Try: true > > > > Bug: webrtc:9419 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103740 > > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#24980} > > > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > > > Change-Id: I4b7cfe492f2c8eeda5c8ac52520e0cfc95ade9b0 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: webrtc:9419 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103801 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#24983} > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9419 > > Change-Id: Id986a0a03cdc2818690337784396882af067f7fa > > Reviewed-on: https://webrtc-review.googlesource.com/c/104602 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25049} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > Change-Id: I6f58b9c90defccdb160307783fb55271ab424fa1 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9419 > Reviewed-on: https://webrtc-review.googlesource.com/c/104623 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25050} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org Change-Id: I4d01ed96ae40a8f9ca42c466be5c87653d75d7c1 Bug: webrtc:9419 Reviewed-on: https://webrtc-review.googlesource.com/c/104641 Reviewed-by: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25108}
2018-10-11 10:50:45 +02:00
"../../rtc_base/system:rtc_export",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:cpu_features_api",
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:field_trial",
"../../system_wrappers:metrics",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"aec3",
Split aec and aecm into separate build targets This clarifies dependencies and makes it easier to customize builds for different binaries. Also adds BUILD files in aec/ and aecm/. Moves unit tests to their own target, which subjects them to Chromium Clang style checks. The CL contains a fix for a thusly induced warning. Bug: webrtc:9488 Change-Id: I77b680b42a4dccc5f025005e0890f60b4eaf2961 Reviewed-on: https://webrtc-review.googlesource.com/87304 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23887}
2018-07-09 14:28:15 +02:00
"aecm:aecm_core",
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
"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
"agc:gain_control_interface",
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
"agc:legacy_agc",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
"agc2:adaptive_digital",
"agc2:fixed_digital",
Activate the pre-amplifier in AudioProcessing. It's a module for applying a gain to the capture signal. The gain is the first processing step in APM. After this CL, these two features work: * The PreAmplifier can be activated with AudioProcessing::Config::pre_amplifier * The PreApmlifier can be controlled after APM creation by AudioProcessing::SetRuntimeSetting. What's left is a change to AecDumps and to AecDump-replay. NOTRY=True # 1-line change, tests just passed. Bug: webrtc:9138 Change-Id: I85b3af511695b0a9cec2eed6fee7f05080305e1d Reviewed-on: https://webrtc-review.googlesource.com/69811 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22881}
2018-04-16 16:31:22 +02:00
"agc2:gain_applier",
Reland "Refactoring of the noise suppressor and adding true multichannel support" This is a reland of 87a7b82520b83a6cf42da27cdc46142c2eb6248c Original change's description: > Refactoring of the noise suppressor and adding true multichannel support > > This CL adds proper multichannel support to the noise suppressor. > To accomplish that in a safe way, a full refactoring of the noise > suppressor code has been done. > > Due to floating point precision, the changes made are not entirely > bitexact. They are, however, very close to being bitexact. > > As a safety measure, the former noise suppressor code is preserved > and a kill-switch is added to allow revering to the legacy noise > suppressor in case issues arise. > > Bug: webrtc:10895, b/143344262 > Change-Id: I0b071011b23265ac12e6d4b3956499d122286657 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407 > Commit-Queue: Per Åhgren <peah@webrtc.org> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29646} Bug: webrtc:10895, b/143344262 Change-Id: I236f1e67bb0baa4e30908a4cf7a8a7bb55fbced3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158747 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29663}
2019-10-29 22:59:44 +01:00
"ns",
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
"transient:transient_suppressor",
Made modules/audio_processing/vad its own target. WHAT: made a BUILD.gn with library and tests in the Audio Processing Module Voice Activity Detector directory. Updated depending code. Fixed a Clang warning. WHY: to make it possible for a target to depend on just the VAD and not the whole APM. There are other benefits: * Sometimes faster compilation. * The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making a peerconnection shared object file without the VAD has to be done in steps. The first step is a custom target for the VAD. Hence this Cl. Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af BUG: webrtc:5716, webrtc:7494 Reviewed-on: https://webrtc-review.googlesource.com/47521 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21893}
2018-02-05 11:15:23 +01:00
"vad",
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 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
deps += [
"../../common_audio",
Fix circular deps in common_audio. This makes it easier to import cleanly downstream, and makes it a lot easier to reason about. Bug: webrtc:6828 Change-Id: I953e129de73053f8619333fe7e318b36e3a1fffa Reviewed-on: https://webrtc-review.googlesource.com/22722 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20727}
2017-11-17 11:04:15 +01:00
"../../common_audio:fir_filter",
"../../common_audio:fir_filter_factory",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"../../system_wrappers",
]
}
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
rtc_library("voice_detection") {
sources = [
"voice_detection.cc",
"voice_detection.h",
]
deps = [
":api",
":audio_buffer",
"../../api/audio:audio_frame_api",
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
]
}
Add empty placeholder target for rms_level.* A follow-up CL will move the rms_level.* files into the new target. Bug: webrtc:11226 Change-Id: I59579b026346f627c0a2739d25f90c12bffbf248 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171102 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30840}
2020-03-20 11:30:32 +01:00
rtc_source_set("rms_level") {
visibility = [ "*" ]
Split RMS level measurement utility from APM This moves rms_level.* into a separate build target. Bug: webrtc:11226 Change-Id: I94ceacd1ec65dda48f5d19b22ba2625d13543e08 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170323 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30856}
2020-03-20 15:55:29 +01:00
sources = [
"rms_level.cc",
"rms_level.h",
]
deps = [
"../../api:array_view",
"../../rtc_base:checks",
"//third_party/abseil-cpp/absl/types:optional",
]
Add empty placeholder target for rms_level.* A follow-up CL will move the rms_level.* files into the new target. Bug: webrtc:11226 Change-Id: I59579b026346f627c0a2739d25f90c12bffbf248 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171102 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30840}
2020-03-20 11:30:32 +01:00
}
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("audio_processing_statistics") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
sources = [
"include/audio_processing_statistics.cc",
"include/audio_processing_statistics.h",
]
deps = [
Reland "Export symbols needed by the Chromium component build (part 1)." This reverts commit 16fe3f290a524a136f71660a114d0b03ef501f10. Reason for revert: After discussing this problem with nisse@ and yvesg@, we decided to modify how RTC_EXPORT works and avoid to depend on the macro COMPONENT_BUILD. RTC_EXPORT will instead depend on a macro WEBRTC_COMPONENT_BUILD (which can be set as a GN argument which defaults to false). When all the symbols needed by Chromium will be marked with RTC_EXPORT we will flip the GN arg in Chromium, setting to to `component_build` and from that moment, Chromium will depend on a WebRTC shared library when `component_build=true`. Original change's description: > Revert "Export symbols needed by the Chromium component build (part 1)." > > This reverts commit 99eea42fc1fe0be0ebed13c5eba7e1e42059bc5a. > > Reason for revert: > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::UnwrapTurnPacket(unsigned char const *, unsigned int, unsigned int *, unsigned int *)" (__imp_?UnwrapTurnPacket@cricket@@YA_NPBEIPAI1@Z) > >>> referenced by obj/services/network/network_service/socket_manager.obj:("virtual void __thiscall network::P2PSocketManager::DumpPacket(class base::span<unsigned char const, 4294967295>, bool)" (?DumpPacket@P2PSocketManager@network@@EAEXV?$span@$$CBE$0PPPPPPPP@@base@@_N@Z)) > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ValidateRtpHeader(unsigned char const *, unsigned int, unsigned int *)" (__imp_?ValidateRtpHeader@cricket@@YA_NPBEIPAI@Z) > >>> referenced by obj/services/network/network_service/socket_manager.obj:("virtual void __thiscall network::P2PSocketManager::DumpPacket(class base::span<unsigned char const, 4294967295>, bool)" (?DumpPacket@P2PSocketManager@network@@EAEXV?$span@$$CBE$0PPPPPPPP@@base@@_N@Z)) > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ApplyPacketOptions(unsigned char *, unsigned int, struct rtc::PacketTimeUpdateParams const &, unsigned __int64)" (__imp_?ApplyPacketOptions@cricket@@YA_NPAEIABUPacketTimeUpdateParams@rtc@@_K@Z) > >>> referenced by obj/services/network/network_service/socket_tcp.obj:("virtual void __thiscall network::P2PSocketTcp::DoSend(class net::IPEndPoint const &, class std::vector<signed char, class std::allocator<signed char>> const &, struct rtc::PacketOptions const &, struct net::NetworkTrafficAnnotationTag)" (?DoSend@P2PSocketTcp@network@@MAEXABVIPEndPoint@net@@ABV?$vector@CV?$allocator@C@std@@@std@@ABUPacketOptions@rtc@@UNetworkTrafficAnnotationTag@4@@Z)) > >>> referenced by obj/services/network/network_service/socket_tcp.obj:("virtual void __thiscall network::P2PSocketStunTcp::DoSend(class net::IPEndPoint const &, class std::vector<signed char, class std::allocator<signed char>> const &, struct rtc::PacketOptions const &, struct net::NetworkTrafficAnnotationTag)" (?DoSend@P2PSocketStunTcp@network@@MAEXABVIPEndPoint@net@@ABV?$vector@CV?$allocator@C@std@@@std@@ABUPacketOptions@rtc@@UNetworkTrafficAnnotationTag@4@@Z)) > lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ApplyPacketOptions(unsigned char *, unsigned int, struct rtc::PacketTimeUpdateParams const &, unsigned __int64)" (__imp_?ApplyPacketOptions@cricket@@YA_NPAEIABUPacketTimeUpdateParams@rtc@@_K@Z) > >>> referenced by obj/services/network/network_service/socket_udp.obj:("bool __thiscall network::P2PSocketUdp::DoSend(struct network::P2PSocketUdp::PendingPacket const &)" (?DoSend@P2PSocketUdp@network@@AAE_NABUPendingPacket@12@@Z)) > > Original change's description: > > Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" > > > > This reverts commit b49520bfc08f5c5832dda1d642125f0bb898f974. > > > > Reason for revert: Problem fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1261398. > > > > Original change's description: > > > Revert "Reland "Export symbols needed by the Chromium component build (part 1)."" > > > > > > This reverts commit 588f4642d1a29f7beaf28265dbd08728191b4c52. > > > > > > Reason for revert: Breaks WebRTC Chromium FYI Win Builder (dbg). > > > lld-link: error: undefined symbol: "__declspec(dllimport) __thiscall webrtc::Config::Config(void)" (__imp_??0Config@webrtc@@QAE@XZ) > > > [...] > > > > > > Original change's description: > > > > Reland "Export symbols needed by the Chromium component build (part 1)." > > > > > > > > This reverts commit 2ea9af227517556136fd629dd2663c0d75d77c7b. > > > > > > > > Reason for revert: The problem will be fixed by > > > > https://chromium-review.googlesource.com/c/chromium/src/+/1261122. > > > > > > > > Original change's description: > > > > > Revert "Export symbols needed by the Chromium component build (part 1)." > > > > > > > > > > This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53. > > > > > > > > > > Reason for revert: Breaks chromium.webrtc.fyi bots. > > > > > > > > > > Original change's description: > > > > > > Export symbols needed by the Chromium component build (part 1). > > > > > > > > > > > > This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) > > > > > > to mark WebRTC symbols as visible from a shared library, this doesn't > > > > > > mean these symbols are part of the public API (please continue to refer > > > > > > to [1] for info about what is considered public WebRTC API). > > > > > > > > > > > > [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md > > > > > > > > > > > > Bug: webrtc:9419 > > > > > > Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99 > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103505 > > > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > > > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > > > > > Cr-Commit-Position: refs/heads/master@{#24969} > > > > > > > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > > > > > > > Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f > > > > > No-Presubmit: true > > > > > No-Tree-Checks: true > > > > > No-Try: true > > > > > Bug: webrtc:9419 > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103720 > > > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#24974} > > > > > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > > > > > Change-Id: I83bbc7f550fc23e823c4d055e0a6f60c828960dd > > > > No-Presubmit: true > > > > No-Tree-Checks: true > > > > No-Try: true > > > > Bug: webrtc:9419 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103740 > > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#24980} > > > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > > > Change-Id: I4b7cfe492f2c8eeda5c8ac52520e0cfc95ade9b0 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: webrtc:9419 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103801 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#24983} > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9419 > > Change-Id: Id986a0a03cdc2818690337784396882af067f7fa > > Reviewed-on: https://webrtc-review.googlesource.com/c/104602 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25049} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org > > Change-Id: I6f58b9c90defccdb160307783fb55271ab424fa1 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9419 > Reviewed-on: https://webrtc-review.googlesource.com/c/104623 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25050} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org Change-Id: I4d01ed96ae40a8f9ca42c466be5c87653d75d7c1 Bug: webrtc:9419 Reviewed-on: https://webrtc-review.googlesource.com/c/104641 Reviewed-by: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25108}
2018-10-11 10:50:45 +02:00
"../../rtc_base/system:rtc_export",
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 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
]
}
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
rtc_source_set("audio_frame_view") {
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 = [ "include/audio_frame_view.h" ]
deps = [ "../../api:array_view" ]
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
}
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
rtc_source_set("audio_generator_interface") {
visibility = [ "*" ]
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 = [ "include/audio_generator.h" ]
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
deps = [
":audio_frame_view",
"../../rtc_base:rtc_base_approved",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../system_wrappers",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
]
}
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("audio_generator_factory") {
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
visibility = [ "*" ]
Mark the audio_generator_factory target as testonly Bug: None Change-Id: Iad9b740835e373b254c9702f8178b4bb41eb36c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147266 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28706}
2019-07-30 11:55:36 +02:00
testonly = true
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
sources = [
"include/audio_generator_factory.cc",
"include/audio_generator_factory.h",
]
deps = [
":audio_generator_interface",
":file_audio_generator",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
"../../rtc_base:rtc_base_approved",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../system_wrappers",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
]
}
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("file_audio_generator") {
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
sources = [
"audio_generator/file_audio_generator.cc",
"audio_generator/file_audio_generator.h",
]
deps = [
":audio_generator_interface",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
"../../rtc_base:rtc_base_approved",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../system_wrappers",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
]
}
GN: Prefix WebRTC specific variables with "rtc_" BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
if (rtc_enable_protobuf) {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
proto_library("audioproc_debug_proto") {
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 = [ "debug.proto" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Fixing WebRTC after moving from src/webrtc to src/ 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}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_processing"
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
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("apm_logging") {
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
configs += [ ":apm_debug_dump" ]
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
sources = [
"logging/apm_data_dumper.cc",
"logging/apm_data_dumper.h",
]
deps = [
"../../api:array_view",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Add output directory option for audioproc_f data dump files. Bug: webrtc:10000 Change-Id: Iac21f826e78d6cb339c68fdeeedf9fe39920ac31 Reviewed-on: https://webrtc-review.googlesource.com/c/110904 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25713}
2018-11-20 12:44:15 +01:00
"../../rtc_base:checks",
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
"../../rtc_base:rtc_base_approved",
]
defines = []
}
Revert "Fix all circular deps in audio_processing (but one)." This reverts commit 0af8370cb38b0b0f35f4ed4ec4237d0e6c7d59da. Reason for revert: Breaks downstream Original change's description: > Fix all circular deps in audio_processing (but one). > > Arguably we should add a few more targets, for instance a utility > target, but I tried to create as few targets as possible here based on > the current usage. > > Bug: webrtc:6828 > Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a > Reviewed-on: https://webrtc-review.googlesource.com/28020 > Reviewed-by: Per Åhgren <peah@webrtc.org> > Commit-Queue: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21025} TBR=phoglund@webrtc.org,peah@webrtc.org Change-Id: I423f027f6919cf4eb44b4e08c7cb38f0506ad0d7 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:6828 Reviewed-on: https://webrtc-review.googlesource.com/28940 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21027}
2017-12-04 10:11:11 +00:00
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_include_tests) {
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
rtc_source_set("mocks") {
testonly = true
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 = [ "include/mock_audio_processing.h" ]
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
deps = [
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
":api",
Use the AEC3 high-pass filter for the whole APM This CL removes and replaces the legacy fixed-point high-pass filter in APM with the floating point high-pass filter in AEC3. Bug: webrtc:10907 Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28950}
2019-08-23 21:29:17 +02:00
":audio_buffer",
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
":audio_processing",
":audio_processing_statistics",
"../../test:test_support",
]
}
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("audio_processing_tests") {
testonly = true
Stop using public_deps in modules/audio_processing. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: Ib44266389e6f08a77bd92cffd1eba166147687f4 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29822 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21106}
2017-12-05 15:07:16 +01:00
deps = [
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
":audioproc_test_utils",
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
"transient:click_annotate",
"transient:transient_suppression_test",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
]
if (rtc_enable_protobuf) {
Stop using public_deps in modules/audio_processing. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: Ib44266389e6f08a77bd92cffd1eba166147687f4 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29822 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21106}
2017-12-05 15:07:16 +01:00
deps += [
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
":audioproc_unittest_proto",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"aec_dump:aec_dump_unittests",
Reland "C++ porting of the initial python script for conversational speech generation." This CL removes the Python script and adds its C++ porting. The former was in its early stage and it has permanently been removed. This is a reland of https://codereview.webrtc.org/2740063004/ which was reverted. Now the build errors are fixed. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2752793002 Cr-Commit-Position: refs/heads/master@{#17300}
2017-03-18 02:29:13 -07:00
"test/conversational_speech",
Parallel iterator bash script. The Python files are empty ones, just a placeholder for the code they will host. Sample audio files for the APM quality assessment toolbox BUG=webrtc:7218 NOTRY=True Review-Url: https://codereview.webrtc.org/2711923002 Cr-Commit-Position: refs/heads/master@{#16827}
2017-02-24 05:32:21 -08:00
"test/py_quality_assessment",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
]
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
}
}
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("audio_processing_unittests") {
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
configs += [ ":apm_debug_dump" ]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"audio_buffer_unittest.cc",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"audio_frame_view_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"config_unittest.cc",
Reland "Remove APM-internal usage of EchoControlMobile" This is a reland of 2fbb83b16b4c2c1712cbe898ca3ba42d6da3e96f Original change's description: > Remove APM-internal usage of EchoControlMobile > > This is a sibling CL to a similar one for EchoCancellation: > https://webrtc-review.googlesource.com/c/src/+/97603 > > - EchoControlMobileImpl will no longer inherit EchoControlMobile. > - Removes usage of AudioProcessing::echo_control_mobile() inside most of > the audio processing module and unit tests. > > The CL breaks audioproc_f backwards compatibility: It can no longer > use all recorded settings (comfort noise, routing mode), but prints an > error message when unsupported settings are encountered. > > Tested: audioproc_f with .wav and aecdump inputs. > Bug: webrtc:9535 > Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce > Reviewed-on: https://webrtc-review.googlesource.com/101621 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Sam Zackrisson <saza@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24888} Bug: webrtc:9535 Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7 Reviewed-on: https://webrtc-review.googlesource.com/c/102881 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24943}
2018-09-28 12:40:47 +02:00
"echo_control_mobile_unittest.cc",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"gain_controller2_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"splitting_filter_unittest.cc",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"test/fake_recording_device_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
deps = [
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
":analog_mic_simulation",
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
":api",
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
":apm_logging",
Adding a placeholder audio_buffer build target This CL adds a placeholder build target in preparation for an upcoming CL (https://webrtc-review.googlesource.com/c/src/+/125081). Bug: webrtc:10366 Change-Id: I5b226e01d561689acf1624e2c0bad30cc1865011 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125560 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26965}
2019-03-05 08:40:48 +01:00
":audio_buffer",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
":audio_frame_view",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":audio_processing",
":audioproc_test_utils",
Isolating APM API build target: adding dummy :api target. This CL is part of a refactoring work to unblock other CLs that would generate a circular dependency when including modules/audio_processing. It will also allow to easily move the APM interface part under //api. More in detail, this change adds a dummy build target named :api in modules/audio_processing. It is needed to adapt the downstream projects before the APM interface files are moved to the :api target. A follow up CL will make :api an actual target and will remove the interface files from :audio_processing. Bug: webrtc:9535 Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f Reviewed-on: https://webrtc-review.googlesource.com/c/109585 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:24:51 +01:00
":config",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
":file_audio_generator_unittests",
Use the AEC3 high-pass filter for the whole APM This CL removes and replaces the legacy fixed-point high-pass filter in APM with the floating point high-pass filter in AEC3. Bug: webrtc:10907 Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28950}
2019-08-23 21:29:17 +02:00
":high_pass_filter",
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
":mocks",
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
":voice_detection",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:aec3_config",
Move EchoCanceller3Factory to api/auido The AEC3 factory is now part of the WebRTC API. Bug: webrtc:8844 Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e Reviewed-on: https://webrtc-review.googlesource.com/57141 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22204}
2018-02-27 13:58:45 +01:00
"../../api/audio:aec3_factory",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../../rtc_base:rtc_base_tests_utils",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Use TaskQueueForTest instead or TaskQueue in unittests To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue Bug: webrtc:10284 Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:37:01 +01:00
"../../rtc_base:task_queue_for_test",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Move file_wrapper.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I440974da4d347b09ff042478720d7983056b62b9 Reviewed-on: https://webrtc-review.googlesource.com/21226 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 10:39:34 +01:00
"../../rtc_base/system:file_wrapper",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
"../../system_wrappers:cpu_features_api",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Add ability to disable detailed error message in RTC_CHECKs Bug: webrtc:11133 Change-Id: I989654f1fb97b476a17956d69ee374406439ea8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160653 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29952}
2019-11-28 17:09:30 +01:00
"../../test:rtc_expect_death",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:test_support",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../audio_coding:neteq_input_audio_tools",
MockAecDump and integration tests between AecDump and AudioProcessing This CL adds a MockAecDump and integration tests that inject the mock into AudioProcessingImpl. The tests check the call pattern between AudioProcessingImpl and AecDump. The existing tests ApmTest.* and DebugDumpTest.* (not touched by this CL) check that the data written by AecDumpImpl is valid. The tests check that the protobuf-writing methods for the different protobuf message types in audio_processing/debug.proto are indeed called for the different modes of AudioProcessingImpl operation. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2888533005 Cr-Commit-Position: refs/heads/master@{#18501}
2017-06-08 08:12:46 -07:00
"aec_dump:mock_aec_dump_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
"agc:agc_unittests",
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
"agc2:adaptive_digital_unittests",
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"agc2:biquad_filter_unittests",
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
"agc2:fixed_digital_unittests",
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
"agc2:noise_estimator_unittests",
AGC2: adding level estimation option (RMS or peak-based). This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:44:15 +01:00
"agc2:rnn_vad_with_level_unittests",
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 21:31:38 +01:00
"agc2:test_utils",
AGC2 RNN VAD: Sequence buffer The SequenceBuffer class template implements a linear buffer with a Push operation that is used to add a fixed size chunk of new samples into the buffer. Its properties are its size and the size of the chunks that are pushed. It is used to implement the pitch buffer in the RNN VAD feature extractor, for which a ring buffer would be a painful choice. Bug: webrtc:9076 Change-Id: I4767bf06d5a414dbed724a96ea4186ef013a1e30 Reviewed-on: https://webrtc-review.googlesource.com/70204 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22919}
2018-04-17 16:46:45 +02:00
"agc2/rnn_vad:unittests",
Conversational Speech tool, rtc_test target replaced with entry in modules_unittests The ConversationalSpeechTest.* unit tests are now part of modules_unittests. The rtc_test target has been replaced with an rtc_source_set one. The latter is included as dependency in audio_processing_unittests. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2769863005 Cr-Commit-Position: refs/heads/master@{#17360}
2017-03-23 05:17:06 -07:00
"test/conversational_speech:unittest",
Split out some dependencies from the monolith audio processing target This is a first step to make the transient suppressor and voice detection optional. Bug: webrtc:11226, webrtc:11292 Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30783}
2020-03-12 12:45:21 +01:00
"transient:transient_suppression_unittests",
Create separate build targets for utility/ in APM This clarifies the dependencies of utility/ a lot (spoiler: there are very few) and makes it easier to separate the build targets for aecm and aec2. Bug: webrtc:9488 Change-Id: If916f86e80c19d1b650d0908fbe8343ea7c47bd7 Reviewed-on: https://webrtc-review.googlesource.com/87141 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23860}
2018-07-05 16:41:55 +02:00
"utility:legacy_delay_estimator_unittest",
PFFFT C++ wrapper for APM Pretty-Fast Fast Fourier Transform is a 3rd party FFT C library meant to replace other FFT libraries in WebRTC (see https://crbug.com/webrtc/9577). This CL adds a WebRTC wrapper meant to be used inside the Audio Processing Module (APM). As a first step, it only supports aligned memory allocated via PFFFT. Support for the C++ standard library containers will be done afterwards since it requires careful investigation and benchmarking (because PFFFT uses SIMD optimizations). The wrapper pre-allocates a scratch buffer to avoid VLA. Bug: webrtc:9577 Change-Id: Ied00c3d3b1df292024f608ccf0ed1917d6e92e56 Reviewed-on: https://webrtc-review.googlesource.com/c/122563 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Max Morin <maxmorin@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26808}
2019-02-21 17:35:43 +01:00
"utility:pffft_wrapper_unittest",
Made modules/audio_processing/vad its own target. WHAT: made a BUILD.gn with library and tests in the Audio Processing Module Voice Activity Detector directory. Updated depending code. Fixed a Clang warning. WHY: to make it possible for a target to depend on just the VAD and not the whole APM. There are other benefits: * Sometimes faster compilation. * The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making a peerconnection shared object file without the VAD has to be done in steps. The first step is a custom target for the VAD. Hence this Cl. Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af BUG: webrtc:5716, webrtc:7494 Reviewed-on: https://webrtc-review.googlesource.com/47521 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21893}
2018-02-05 11:15:23 +01:00
"vad:vad_unittests",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"//testing/gtest",
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 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
defines = []
if (rtc_prefer_fixed_point) {
defines += [ "WEBRTC_AUDIOPROC_FIXED_PROFILE" ]
} else {
defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ]
}
if (rtc_enable_protobuf) {
defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
deps += [
GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2647003002 Cr-Commit-Position: refs/heads/master@{#16209}
2017-01-23 04:26:02 -08:00
":audioproc_debug_proto",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":audioproc_protobuf_utils",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
":audioproc_test_utils",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":audioproc_unittest_proto",
Split RMS level measurement utility from APM This moves rms_level.* into a separate build target. Bug: webrtc:11226 Change-Id: I94ceacd1ec65dda48f5d19b22ba2625d13543e08 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170323 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30856}
2020-03-20 15:55:29 +01:00
":rms_level",
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 10:18:07 +02:00
":runtime_settings_protobuf_utils",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
APM unit test: echo path gain change events notified. This CL adds two unit tests to make sure that, when an echo path gain change occurs, the echo canceller is notified. Such a change can be caused by (i) a pre-amplifier gain change or (ii) an analog gain change. Bug: webrtc:7494 Change-Id: Ia47cfbbc5694340cd3e760d8d3c3393f79897a9d Reviewed-on: https://webrtc-review.googlesource.com/c/111780 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26190}
2018-12-17 09:44:06 +01:00
"../../api/audio:echo_control",
"../../rtc_base:rtc_base_tests_utils",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_task_queue",
Change existing aec dump tests to use webrtc::AecDump. Currently the debug dump functionality of WebRTC (a log of all AudioProcessing operations) was tested by the following tests: 1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing from a debug dump, and verifies that the same debug dump is recorded. 2. DebugDumpTest.* which is a comprehensive test of the debug dump operations. AudioProcessing configuration is changed, and the dump is scanned for the change. 3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that debug dumping can be started and files written. This CL replaces the debug dump mechanism in all these tests to webrtc::AecDump. Some of the tests are adapted to the chenges of the new API to AecDump {Start,Stop}DebugRecording: the old functions signal errors when a file cannot be opened. With AecDump, the AecDumpFactory instead returns a nullptr. The CL also changes audioproc_f to use AecDump. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2864373002 Cr-Commit-Position: refs/heads/master@{#18605}
2017-06-15 01:55:38 -07:00
"aec_dump",
"aec_dump:aec_dump_unittests",
Reintroduce command line controlled reference data updating for ApmTest.Process Replaces a hardcoded bool in a test with command line flag. The current hardcoding of the bool is a little bit hacky. And the tests will pass automatically, so it is possible to accidentally commit the flipped bool in a CL, like here: https://webrtc-review.googlesource.com/c/src/+/150221 I am fairly sure this resolves the vague issue referred to in the attached bug. The bug is introduced with a TODO here: https://webrtc-codereview.appspot.com/1728005/diff/4001/webrtc/modules/audio_processing/test/unit_test.cc Another TODO was added later that refers to the first TODO: https://webrtc-codereview.appspot.com/6879004/diff/150001/webrtc/modules/audio_processing/test/audio_processing_unittest.cc Bug: webrtc:1981 Change-Id: I066f41add602c791a5f2ba18829c4306da7dac15 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150334 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28955}
2019-08-26 10:12:41 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
sources += [
"audio_processing_impl_locking_unittest.cc",
"audio_processing_impl_unittest.cc",
"audio_processing_unittest.cc",
Reland "Remove APM-internal usage of EchoControlMobile" This is a reland of 2fbb83b16b4c2c1712cbe898ca3ba42d6da3e96f Original change's description: > Remove APM-internal usage of EchoControlMobile > > This is a sibling CL to a similar one for EchoCancellation: > https://webrtc-review.googlesource.com/c/src/+/97603 > > - EchoControlMobileImpl will no longer inherit EchoControlMobile. > - Removes usage of AudioProcessing::echo_control_mobile() inside most of > the audio processing module and unit tests. > > The CL breaks audioproc_f backwards compatibility: It can no longer > use all recorded settings (comfort noise, routing mode), but prints an > error message when unsupported settings are encountered. > > Tested: audioproc_f with .wav and aecdump inputs. > Bug: webrtc:9535 > Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce > Reviewed-on: https://webrtc-review.googlesource.com/101621 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Sam Zackrisson <saza@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24888} Bug: webrtc:9535 Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7 Reviewed-on: https://webrtc-review.googlesource.com/c/102881 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24943}
2018-09-28 12:40:47 +02:00
"echo_control_mobile_bit_exact_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"echo_detector/circular_buffer_unittest.cc",
"echo_detector/mean_variance_estimator_unittest.cc",
"echo_detector/moving_max_unittest.cc",
"echo_detector/normalized_covariance_estimator_unittest.cc",
"gain_control_unittest.cc",
Use the AEC3 high-pass filter for the whole APM This CL removes and replaces the legacy fixed-point high-pass filter in APM with the floating point high-pass filter in AEC3. Bug: webrtc:10907 Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28950}
2019-08-23 21:29:17 +02:00
"high_pass_filter_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"level_estimator_unittest.cc",
"residual_echo_detector_unittest.cc",
"rms_level_unittest.cc",
"test/debug_dump_replayer.cc",
"test/debug_dump_replayer.h",
"test/debug_dump_test.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"test/echo_canceller_test_tools.cc",
"test/echo_canceller_test_tools.h",
"test/echo_canceller_test_tools_unittest.cc",
APM unit test: echo path gain change events notified. This CL adds two unit tests to make sure that, when an echo path gain change occurs, the echo canceller is notified. Such a change can be caused by (i) a pre-amplifier gain change or (ii) an analog gain change. Bug: webrtc:7494 Change-Id: Ia47cfbbc5694340cd3e760d8d3c3393f79897a9d Reviewed-on: https://webrtc-review.googlesource.com/c/111780 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26190}
2018-12-17 09:44:06 +01:00
"test/echo_control_mock.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"test/test_utils.h",
"voice_detection_unittest.cc",
]
}
}
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("audio_processing_perf_tests") {
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
testonly = true
Delete ApmPublicSubmodules, fix ApmPrivateSubmodules, ApmSubmoduleStates All submodule pointers are now private. The unique_ptr to a ApmPrivateSubmodules is replaced by a direct member object. The main outcome of this CL is that the code is nicer. Bug: webrtc:5298 Change-Id: Ib8ef70a35a64b875752d2a318c572d152d51487a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157440 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29539}
2019-10-18 13:29:43 +02:00
configs += [ ":apm_debug_dump" ]
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
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 = [ "audio_processing_performance_unittest.cc" ]
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
deps = [
":audio_processing",
":audioproc_test_utils",
Removed residual echo complexity unittest. This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it. Bug: chromium:788318 Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde Reviewed-on: https://webrtc-review.googlesource.com/25824 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 15:35:57 +01:00
"../../api:array_view",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
Removed residual echo complexity unittest. This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it. Bug: chromium:788318 Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde Reviewed-on: https://webrtc-review.googlesource.com/25824 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 15:35:57 +01:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../../test:perf_test",
Removed residual echo complexity unittest. This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it. Bug: chromium:788318 Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde Reviewed-on: https://webrtc-review.googlesource.com/25824 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 15:35:57 +01:00
"../../test:test_support",
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
]
}
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("file_audio_generator_unittests") {
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
testonly = true
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 = [ "audio_generator/file_audio_generator_unittest.cc" ]
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
deps = [
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
":api",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
":audio_generator_factory",
":audio_processing",
":file_audio_generator",
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
"../../test:test_support",
]
}
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("analog_mic_simulation") {
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
sources = [
"test/fake_recording_device.cc",
"test/fake_recording_device.h",
]
deps = [
"../../api:array_view",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"../../rtc_base:rtc_base_approved",
Add AGC1-compliant fake recording device. The AGC submodule of APM changes analog gain. These gain changes are typically ignored by the test tool audioproc_f. There is an option of the test tool to take action on the gain changes. It's the '--simulate_mic_gain' option. The option converts the analog gain to a digital gain. The digital gain is applied to the capture stream. This change adds a new simulated microphone kind. The new microphone has a gain curve defined by modules/audio_processing/agc/gain_map_internal.h. That gain curve defines how AGC1 expects a microphone to behave. Bug: webrtc:7494 Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780 Reviewed-on: https://webrtc-review.googlesource.com/86128 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23801}
2018-07-02 11:37:47 +02:00
"../../rtc_base:safe_minmax",
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
"agc:gain_map",
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 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
]
}
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
if (rtc_enable_protobuf) {
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("audioproc_f_impl") {
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
testonly = true
APM: Add ability to turn on/off dumping of internal data This CL modifies the internal data logging and the audioproc_f tool to allow controlling that via the command line, rather than solely via a build flag. The logging of internal data is by default off. Bug: webrtc:5298 Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca Reviewed-on: https://webrtc-review.googlesource.com/c/107713 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25352}
2018-10-25 09:56:49 +02:00
configs += [ ":apm_debug_dump" ]
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
sources = [
"test/aec_dump_based_simulator.cc",
"test/aec_dump_based_simulator.h",
Added more refined benchmarking code for audioproc_f This CL extends, and partly corrects, the benchmarking code in audioproc_f to provide statistics for the API call durations in audioproc_f Bug: chromium:939791 Change-Id: I4c26c4bb3782335f13dd3e21e6f861842539ea62 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129260 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27443}
2019-04-03 16:06:42 +02:00
"test/api_call_statistics.cc",
"test/api_call_statistics.h",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
"test/audio_processing_simulator.cc",
"test/audio_processing_simulator.h",
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 12:22:52 +01:00
"test/audioproc_float_impl.cc",
"test/audioproc_float_impl.h",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
"test/wav_based_simulator.cc",
"test/wav_based_simulator.h",
]
Adding GN targets for audioproc, audioproc_f and unpack_aecdump BUG=webrtc:6171 NOTRY=True Review-Url: https://codereview.webrtc.org/2222643002 Cr-Commit-Position: refs/heads/master@{#13667}
2016-08-07 23:19:30 -07:00
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
deps = [
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
":analog_mic_simulation",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
":api",
APM: Add ability to turn on/off dumping of internal data This CL modifies the internal data logging and the audioproc_f tool to allow controlling that via the command line, rather than solely via a build flag. The logging of internal data is by default off. Bug: webrtc:5298 Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca Reviewed-on: https://webrtc-review.googlesource.com/c/107713 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25352}
2018-10-25 09:56:49 +02:00
":apm_logging",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
":audio_processing",
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 10:18:07 +02:00
":runtime_settings_protobuf_utils",
Add JSON parsing and corresponding ToString to EchoCanceller3Config Bug: webrtc:9535 Change-Id: I51eaaac4009a30536444292a32938b21e69386bf Reviewed-on: https://webrtc-review.googlesource.com/c/102980 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25083}
2018-10-10 10:44:43 +02:00
"../../api/audio:aec3_config_json",
Move EchoCanceller3Factory to api/auido The AEC3 factory is now part of the WebRTC API. Bug: webrtc:8844 Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e Reviewed-on: https://webrtc-review.googlesource.com/57141 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22204}
2018-02-27 13:58:45 +01:00
"../../api/audio:aec3_factory",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Extend the audioproc_f input parameters to match what is supported by AEC3 This CL extends the options for the audioproc_f tool to match the options for AEC3. Bug: webrtc:8671 Change-Id: I39972eae33dba461b94118ec47a8560eb9cfe5a6 Reviewed-on: https://webrtc-review.googlesource.com/43120 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22344}
2018-03-08 07:47:14 +01:00
"../../rtc_base:rtc_json",
Use TaskQueueForTest instead or TaskQueue in unittests To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue Bug: webrtc:10284 Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:37:01 +01:00
"../../rtc_base:task_queue_for_test",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
"../../system_wrappers",
"../../test:test_support",
Change existing aec dump tests to use webrtc::AecDump. Currently the debug dump functionality of WebRTC (a log of all AudioProcessing operations) was tested by the following tests: 1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing from a debug dump, and verifies that the same debug dump is recorded. 2. DebugDumpTest.* which is a comprehensive test of the debug dump operations. AudioProcessing configuration is changed, and the dump is scanned for the change. 3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that debug dumping can be started and files written. This CL replaces the debug dump mechanism in all these tests to webrtc::AecDump. Some of the tests are adapted to the chenges of the new API to AecDump {Start,Stop}DebugRecording: the old functions signal errors when a file cannot be opened. With AecDump, the AecDumpFactory instead returns a nullptr. The CL also changes audioproc_f to use AecDump. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2864373002 Cr-Commit-Position: refs/heads/master@{#18605}
2017-06-15 01:55:38 -07:00
"aec_dump",
"aec_dump:aec_dump_impl",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
"//testing/gtest",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
APM audioproc_f: flag for AGC2 adaptive level estimator. Bug: webrtc:7494 Change-Id: I603211570a0a46d8884749dab887cd572827cca6 Reviewed-on: https://webrtc-review.googlesource.com/c/110250 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25708}
2018-11-20 12:27:20 +01:00
"//third_party/abseil-cpp/absl/strings",
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 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
]
Restructure the audioproc_f tool into a library with a thin executable wrapper. This refactoring makes it easier to experiment with injectable components. Bug: webrtc:8732 Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217 Reviewed-on: https://webrtc-review.googlesource.com/60863 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22372}
2018-03-09 17:33:04 +01:00
} # audioproc_f_impl
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
}
Adding GN targets for audioproc, audioproc_f and unpack_aecdump BUG=webrtc:6171 NOTRY=True Review-Url: https://codereview.webrtc.org/2222643002 Cr-Commit-Position: refs/heads/master@{#13667}
2016-08-07 23:19:30 -07:00
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("audioproc_test_utils") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
sources = [
Add webrtc_perf_tests to BUILD.gn Updated the sources in audio_processing:audioproc_test_utils to match the configuration on "webrtc/modules/audio_processing/audio_processing_tests.gypi" Removed audio_buffer_tools from modules_unittests to match the gyp file. BUG=webrtc:6041 Review-Url: https://codereview.webrtc.org/2178963002 Cr-Commit-Position: refs/heads/master@{#13541}
2016-07-27 08:14:32 -07:00
"test/audio_buffer_tools.cc",
"test/audio_buffer_tools.h",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
"test/bitexactness_tools.cc",
"test/bitexactness_tools.h",
Added a perf test for the residual echo detector. This perf tests the echo detector in 3 scenarios: standalone, as part of APM with only the echo detector enabled and as part of a normally configured APM. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2517523003 Cr-Commit-Position: refs/heads/master@{#15224}
2016-11-24 04:17:28 -08:00
"test/performance_timer.cc",
"test/performance_timer.h",
"test/simulator_buffers.cc",
"test/simulator_buffers.h",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"test/test_utils.cc",
"test/test_utils.h",
]
deps = [
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
":api",
Adding a placeholder audio_buffer build target This CL adds a placeholder build target in preparation for an upcoming CL (https://webrtc-review.googlesource.com/c/src/+/125081). Bug: webrtc:10366 Change-Id: I5b226e01d561689acf1624e2c0bad30cc1865011 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125560 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26965}
2019-03-05 08:40:48 +01:00
":audio_buffer",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
":audio_processing",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
"../../test:test_support",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../audio_coding:neteq_input_audio_tools",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
"//testing/gtest",
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 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
}
if (rtc_enable_protobuf) {
proto_library("audioproc_unittest_proto") {
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 = [ "test/unittest.proto" ]
Fixing WebRTC after moving from src/webrtc to src/ 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}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_processing/test"
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
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("audioproc_protobuf_utils") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
sources = [
"test/protobuf_utils.cc",
"test/protobuf_utils.h",
]
deps = [
GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets. When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled. BUG=webrtc:6215 NOTRY=True Review-Url: https://codereview.webrtc.org/2250893003 Cr-Commit-Position: refs/heads/master@{#13829}
2016-08-19 02:11:07 -07:00
":audioproc_debug_proto",
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 10:18:07 +02:00
"../../rtc_base:checks",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
}
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 10:18:07 +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("runtime_settings_protobuf_utils") {
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 10:18:07 +02:00
testonly = true
sources = [
"test/runtime_setting_util.cc",
"test/runtime_setting_util.h",
]
deps = [
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
":api",
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 10:18:07 +02:00
":audioproc_debug_proto",
":audioproc_protobuf_utils",
"../../rtc_base:checks",
]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
}
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