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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")
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
Reland of Fixing package-boundary violation with srjar_deps (patchset #1 id:1 of https://codereview.webrtc.org/2617533005/ ) Reason for revert: Skipping the build of the target "//webrtc/modules/audio_device:audio_device_java" when building webrtc with chromium. This was causing a failure because in that case it is not possible to refer to the script used by the template "android_shared_srcjar" with an absolute path (which seems to be necessary since the script attribute of action is relative to the invocation target and not to the template declaration). Original issue's description: > Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ ) > > Reason for revert: > This CL is breaking a chromium.webrtc.fyi: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/226 > > I am trying to reproduce the issue on my local machine and I will try to re-land the CL later. > > Original issue's description: > > Fixing package-boundary violation with srjar_deps > > > > Without the usage of the srcjar_deps attribute we were not able to > > include .java files from other packages without violating the package > > boundary contraint. > > > > As an example, in this CL the target "libjingle_peerconnection_java" was > > directly including .java files from another packages in its "java_files" > > attribute. > > > > Using srcjar_deps we are able to declare the dependency of the target > > avoiding to create hidden dependencies in the codebase. > > > > This is not fixing the webrtc:6356 bug directly but it is a first step to > > include ThreadUtils classes in libjingle_peerconnection_client_java.jar > > again. > > > > It seems also to be related to the chromium:648244 bug. This can be solved > > if we can find a way to perform srcjar generation in the android_library > > target without changing the semantic of the target. > > > > BUG=webrtc:6356 > > > > Review-Url: https://codereview.webrtc.org/2610823002 > > Cr-Commit-Position: refs/heads/master@{#15914} > > Committed: https://chromium.googlesource.com/external/webrtc/+/10a76592a78762187bfaf3547b7e9986454062ab > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6356 > > Review-Url: https://codereview.webrtc.org/2617533005 > Cr-Commit-Position: refs/heads/master@{#15915} > Committed: https://chromium.googlesource.com/external/webrtc/+/eb731ed09ebfea1b26d1b7bc1f2678f7bbc012ee TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6356 Review-Url: https://codereview.webrtc.org/2612953004 Cr-Commit-Position: refs/heads/master@{#15958}
2017-01-09 02:37:21 -08:00
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
GN: Fix windows clang errors. Attempt 2. BUG=webrtc:6255 NOTRY=True Review-Url: https://codereview.webrtc.org/2281513002 Cr-Commit-Position: refs/heads/master@{#13942}
2016-08-26 13:31:24 -07:00
config("audio_device_warnings_config") {
if (is_win && is_clang) {
cflags = [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
"-Wno-microsoft-goto",
]
}
}
Adding missing build target for audio_device_default. The header modules/audio_device/include/audio_device_default.h was not owned by any build target. Bug: webrtc:8946 Change-Id: I3266a613c10963688c3bea701384e1d1bb68daac Reviewed-on: https://webrtc-review.googlesource.com/c/111201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25669}
2018-11-16 11:13:32 +01:00
rtc_source_set("audio_device_default") {
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_device_default.h" ]
deps = [ ":audio_device_api" ]
Adding missing build target for audio_device_default. The header modules/audio_device/include/audio_device_default.h was not owned by any build target. Bug: webrtc:8946 Change-Id: I3266a613c10963688c3bea701384e1d1bb68daac Reviewed-on: https://webrtc-review.googlesource.com/c/111201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25669}
2018-11-16 11:13:32 +01:00
}
Decoupling audio_device from Obj-C code The goal of this CL is to separate Obj-C/Obj-C++ code from targets which have also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743 for more information). To achieve this we have created 2 targets (audio_device_ios_objc and audio_device_generic) and audio_device will act as a proxy between these targets (this way we can avoid a circular dependency between audio_device_generic and audio_device_ios_objc). BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2991343002 Cr-Commit-Position: refs/heads/master@{#19795}
2017-09-12 04:45:24 -07:00
rtc_source_set("audio_device") {
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 = [ "*" ]
Decoupling audio_device from Obj-C code The goal of this CL is to separate Obj-C/Obj-C++ code from targets which have also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743 for more information). To achieve this we have created 2 targets (audio_device_ios_objc and audio_device_generic) and audio_device will act as a proxy between these targets (this way we can avoid a circular dependency between audio_device_generic and audio_device_ios_objc). BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2991343002 Cr-Commit-Position: refs/heads/master@{#19795}
2017-09-12 04:45:24 -07:00
public_deps = [
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
":audio_device_api",
# Deprecated.
# TODO(webrtc:7452): Remove this public dep. audio_device_impl should
# be depended on directly if needed.
":audio_device_impl",
Decoupling audio_device from Obj-C code The goal of this CL is to separate Obj-C/Obj-C++ code from targets which have also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743 for more information). To achieve this we have created 2 targets (audio_device_ios_objc and audio_device_generic) and audio_device will act as a proxy between these targets (this way we can avoid a circular dependency between audio_device_generic and audio_device_ios_objc). BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2991343002 Cr-Commit-Position: refs/heads/master@{#19795}
2017-09-12 04:45:24 -07:00
]
}
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
rtc_source_set("audio_device_api") {
Moving iOS Audio Device to sdk. This change forks the existing iOS audio device module and audio device from modules/audio_device/ into sdk/objc/Framework. It also updates RTCPeerConnectionFactory to use the forked implementation. The unit tests are re-implemented as XCTests. (was: https://webrtc-review.googlesource.com/c/src/+/67300) Bug: webrtc:9120 Change-Id: I46c09900246f75ca5285aeb38f7b8b295784ffac Reviewed-on: https://webrtc-review.googlesource.com/76741 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Anders Carlsson <andersc@webrtc.org> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23238}
2018-05-15 10:22:36 +02:00
visibility = [ "*" ]
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
sources = [
"include/audio_device.h",
"include/audio_device_defines.h",
]
deps = [
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",
Propagate TaskQueueFactory to AudioDeviceBuffer keep using GlobalTaskQueueFactory in android/ios bindings. Switch to DefaultTaskQueueFactory in tests. Bug: webrtc:10284 Change-Id: I034c70542be5eeb830be86527830d51204fb2855 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 09:16:12 +02:00
"../../api/task_queue",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
Add support of AAudio in native WebRTC on Android O and above Bug: webrtc:8914 Change-Id: I016dd8fcebba1644c0a83e5f1460520545d4cdde Reviewed-on: https://webrtc-review.googlesource.com/56180 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22467}
2018-03-16 10:09:49 +01:00
"../../rtc_base:stringutils",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +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_device_buffer") {
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
"audio_device_config.h",
"fine_audio_buffer.cc",
"fine_audio_buffer.h",
]
deps = [
":audio_device_api",
"../../api:array_view",
Propagate TaskQueueFactory to AudioDeviceBuffer keep using GlobalTaskQueueFactory in android/ios bindings. Switch to DefaultTaskQueueFactory in tests. Bug: webrtc:10284 Change-Id: I034c70542be5eeb830be86527830d51204fb2855 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 09:16:12 +02:00
"../../api/task_queue",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
Migrate modules/audio_device to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I6d1a7145aaaae2e4cd0c8658fa31a673f857dbd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178814 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31664}
2020-07-08 10:13:20 +02:00
"../../rtc_base/synchronization:mutex",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:metrics",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +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_device_generic") {
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
sources = [
"audio_device_generic.cc",
"audio_device_generic.h",
]
deps = [
":audio_device_api",
":audio_device_buffer",
"../../rtc_base:rtc_base_approved",
]
}
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_device_name") {
Ports CoreAudioUtil from Chrome to WebRTC. See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34 for details. Bug: webrtc:9265 Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb Reviewed-on: https://webrtc-review.googlesource.com/76900 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:20:58 +02:00
sources = [
"audio_device_name.cc",
"audio_device_name.h",
]
}
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
rtc_source_set("windows_core_audio_utility") {
if (is_win && !build_with_chromium) {
sources = [
"win/core_audio_utility_win.cc",
"win/core_audio_utility_win.h",
]
deps = [
":audio_device_api",
":audio_device_name",
"../../api/units:time_delta",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
Add a missing Windows library "oleaut32.lib" is required for VariantInit: https://docs.microsoft.com/en-us/windows/win32/api/oleauto/nf-oleauto-variantinit Bug: webrtc:11807 Change-Id: If0511571340e14407ad9402636a4a64d328fabca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180440 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Eric Astor <epastor@google.com> Cr-Commit-Position: refs/heads/master@{#31806}
2020-07-28 17:55:21 -04:00
libs = [ "oleaut32.lib" ]
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
}
}
# An ADM with a dedicated factory method which does not depend on the
# audio_device_impl target. The goal is to use this new structure and
# gradually phase out the old design.
# TODO(henrika): currently only supported on Windows.
rtc_source_set("audio_device_module_from_input_and_output") {
Make audio_device_module_from_input_and_output visible. Bug: None Change-Id: I185fe3d3f4b96fd55fd7148c33df105caa21d9ab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132010 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27511}
2019-04-09 11:32:18 +02:00
visibility = [ "*" ]
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
if (is_win && !build_with_chromium) {
sources = [
"include/audio_device_factory.cc",
"include/audio_device_factory.h",
]
sources += [
"win/audio_device_module_win.cc",
"win/audio_device_module_win.h",
"win/core_audio_base_win.cc",
"win/core_audio_base_win.h",
"win/core_audio_input_win.cc",
"win/core_audio_input_win.h",
"win/core_audio_output_win.cc",
"win/core_audio_output_win.h",
]
deps = [
":audio_device_api",
":audio_device_buffer",
":windows_core_audio_utility",
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",
Propagate TaskQueueFactory to AudioDeviceBuffer keep using GlobalTaskQueueFactory in android/ios bindings. Switch to DefaultTaskQueueFactory in tests. Bug: webrtc:10284 Change-Id: I034c70542be5eeb830be86527830d51204fb2855 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 09:16:12 +02:00
"../../api/task_queue",
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
}
}
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
# Contains default implementations of webrtc::AudioDeviceModule for Windows,
# Linux, Mac, iOS and Android.
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_device_impl") {
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
visibility = [ "*" ]
Add RTCAudioSession proxy class. BUG= R=haysc@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1709853002 . Cr-Commit-Position: refs/heads/master@{#11676}
2016-02-18 15:44:07 -08:00
deps = [
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
":audio_device_api",
":audio_device_buffer",
Adding missing build target for audio_device_default. The header modules/audio_device/include/audio_device_default.h was not owned by any build target. Bug: webrtc:8946 Change-Id: I3266a613c10963688c3bea701384e1d1bb68daac Reviewed-on: https://webrtc-review.googlesource.com/c/111201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25669}
2018-11-16 11:13:32 +01:00
":audio_device_default",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
":audio_device_generic",
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",
Use scoped_refptr to share the instance of OpenSLEngineManager Use rtc::scoped_refptr instead of std::unique_ptr to hold the instance of OpenSLEngineManager; this makes it safe to share it between OpenSLESRecorder and OpenSLESPlayer. Bug: webrtc:10436 Change-Id: Ibd0717e5410020c89a40bfdb05953a02378a6a4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128651 Commit-Queue: Lu Liu <lliuu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27253}
2019-03-22 12:40:05 -07:00
"../../api:refcountedbase",
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",
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
"../../api/task_queue",
Add RTCAudioSession proxy class. BUG= R=haysc@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1709853002 . Cr-Commit-Position: refs/heads/master@{#11676}
2016-02-18 15:44:07 -08:00
"../../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 voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +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:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
Migrate modules/audio_device to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I6d1a7145aaaae2e4cd0c8658fa31a673f857dbd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178814 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31664}
2020-07-08 10:13:20 +02:00
"../../rtc_base/synchronization:mutex",
Remove check_includes=false from audio_device_impl. Bug: webrtc:8850, 824831 Change-Id: I89f089e09214ef189fca97c3f1062a0cd8476687 Reviewed-on: https://webrtc-review.googlesource.com/97622 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24548}
2018-09-04 11:06:17 +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",
Reland "Replacing rtc::Thread with task queue for TestAudioDeviceModule." This is a reland of 1b871d07532c25d2f27e4db192cb9ce2229b1cee Original change's description: > Replacing rtc::Thread with task queue for TestAudioDeviceModule. > > This prepares for running it in simulated time. > > TBR=henrika@webrtc.org > > Bug: webrtc:10465 > Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27083} TBR=henrika@webrtc.org Bug: webrtc:10465 Change-Id: Icda8043fb5b1156129bc3b706bf8f190782b0921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127520 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27093}
2019-03-13 08:50:42 +01:00
"../../rtc_base/task_utils:repeating_task",
Add RTCAudioSession proxy class. BUG= R=haysc@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1709853002 . Cr-Commit-Position: refs/heads/master@{#11676}
2016-02-18 15:44:07 -08:00
"../../system_wrappers",
Allows configuration of playout audio buffer Playout audio buffer length in Java audio device configuration with fieldtrial. Bug: webrtc:10928 Change-Id: I79286f09591f4b2c6a6146f23d3dce92a29f6b21 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150657 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Alex Narest <alexnarest@google.com> Cr-Commit-Position: refs/heads/master@{#29005}
2019-08-29 13:17:11 +02:00
"../../system_wrappers:field_trial",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:metrics",
Add RTCAudioSession proxy class. BUG= R=haysc@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1709853002 . Cr-Commit-Position: refs/heads/master@{#11676}
2016-02-18 15:44:07 -08:00
"../utility",
]
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
if (rtc_include_internal_audio_device && is_ios) {
Reland "Remove old audio device implementation." This reverts commit e9d2b4efdd5dddaa3a476c0ac2a9cf9125b39929. Reason for revert: Fixing build break of downstream project. Original change's description: > Revert "Remove old audio device implementation." > > This reverts commit 0cfa4cba5cae5e942f5d8e0e4e93b94982d0bfc3. > > Reason for revert: audio_device_ios_objc target is removed, but still referenced by iPhone Meetins:Meeting_build_test, which now fails to build > > Original change's description: > > Remove old audio device implementation. > > > > The iOS ADM implementation now lives in sdk/objc/native/api/audio_device_module.{h,mm}. > > > > Bug: webrtc:10514 > > Change-Id: Ib0b162027b5680ebc40d621a57f1155f08e7a057 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131326 > > Commit-Queue: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27488} > > TBR=henrika@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I5be10b3d17403a79ea30afc255cde01171bc9f5b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10514 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131960 > Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org> > Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27492} TBR=henrika@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org,jeroendb@webrtc.org Change-Id: Ic05131d902f9623bd5ae7635d3c71880ffd3c6d3 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10514 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131944 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27533}
2019-04-10 08:43:53 +02:00
deps += [ "../../sdk:audio_device" ]
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
}
Add RTCAudioSession proxy class. BUG= R=haysc@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1709853002 . Cr-Commit-Position: refs/heads/master@{#11676}
2016-02-18 15:44:07 -08:00
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
sources = [
"dummy/audio_device_dummy.cc",
"dummy/audio_device_dummy.h",
"dummy/file_audio_device.cc",
"dummy/file_audio_device.h",
Add missing audio_device .h files. Bug: webrtc:7650 Change-Id: Ifbfcac49da9f761f666c6ff22b934e3a3ab783fd Reviewed-on: https://webrtc-review.googlesource.com/37003 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21482}
2018-01-02 15:00:11 +01:00
"include/fake_audio_device.h",
Cleanup after moving test/fake_audio_device. Cleanup after moving test/fake_audio_device to modules/audio_device/include/test_audio_device. Hide implementation of test audio device module in the anonymous namespace. Bug: webrtc:8946 Change-Id: I2d49c3ec5d43eeb5f155d38de95f69ed3c537805 Reviewed-on: https://webrtc-review.googlesource.com/61426 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22401}
2018-03-13 13:25:19 +01:00
"include/test_audio_device.cc",
Separate test/fake_audio_device on API and implementation. Step 1. Adding ability of injecting audio in end to end tests, that are using WebRTC. It will be done in 3 steps: 1. Test/fake_audio_device will be moved to production part of WebRTC source code and renamed to test_audio_device_module. Old header is replaced with alias to the new one. 2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice. 3. test/fake_audio_device will be removed. This CL implements 1st step. Bug: webrtc:8946 Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f Reviewed-on: https://webrtc-review.googlesource.com/59740 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22325}
2018-03-07 12:20:51 +01:00
"include/test_audio_device.h",
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
]
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
if (build_with_mozilla) {
sources += [
"opensl/single_rw_fifo.cc",
"opensl/single_rw_fifo.h",
]
}
Roll chromium_revision c089d37..159828f (353662:353696) Due to https://codereview.chromium.org/1397493004 we're now adding a build_overrides directory in WebRTC. Thanks to this, we no longer need to pass --args="build_with_chromium=false" when running GN in standalone WebRTC. Change log: https://chromium.googlesource.com/chromium/src/+log/c089d37..159828f Full diff: https://chromium.googlesource.com/chromium/src/+/c089d37..159828f No dependencies changed. No update to Clang. BUG=webrtc:5070,chromium:541791 TBR=tommi@webrtc.org CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal Review URL: https://codereview.webrtc.org/1403453003 . Cr-Commit-Position: refs/heads/master@{#10270}
2015-10-14 08:13:58 +02:00
defines = []
GN: Fix iOS error in audio_device and rtc_base With this in, the only compilation errors left seems related to yasm and libjpeg_turbo. Notice the below example builds x86 builds (not ARM) since if specifying target_cpu="arm", the gn step fails (separate issue). BUG=webrtc:5213, webrtc:5195, chromium:459705 TESTED=Passing compilation with: gn gen --args="target_os=\"ios\"" out/Default ninja -C out/Default rtc_base audio_device Review URL: https://codereview.webrtc.org/1471663002 Cr-Commit-Position: refs/heads/master@{#10763}
2015-11-23 17:23:44 -08:00
cflags = []
Adds detection of audio glitches for playout on iOS (reland) Second attempt to land https://chromium-review.googlesource.com/c/522563/ TBR: minyue Bug: b/38018041 Change-Id: I938f4a490b6357cd1ac7b34fe445215a746fab43 Reviewed-on: https://chromium-review.googlesource.com/533214 Commit-Queue: Henrik Andreasson <henrika@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18572}
2017-06-13 17:34:16 +02:00
if (rtc_audio_device_plays_sinus_tone) {
defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
}
Add support of AAudio in native WebRTC on Android O and above Bug: webrtc:8914 Change-Id: I016dd8fcebba1644c0a83e5f1460520545d4cdde Reviewed-on: https://webrtc-review.googlesource.com/56180 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22467}
2018-03-16 10:09:49 +01:00
if (rtc_enable_android_aaudio) {
Prefix AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO with WEBRTC_. Since it is a WebRTC-only macro, let's prefix it with WEBRTC_. Bug: None Change-Id: I309666858ea898dc7cd1a68c21be190f98c87b11 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129935 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27327}
2019-03-27 21:11:17 +01:00
defines += [ "WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO" ]
Add support of AAudio in native WebRTC on Android O and above Bug: webrtc:8914 Change-Id: I016dd8fcebba1644c0a83e5f1460520545d4cdde Reviewed-on: https://webrtc-review.googlesource.com/56180 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22467}
2018-03-16 10:09:49 +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_include_internal_audio_device) {
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
sources += [
Adding AudioDeviceDataObserver interface Added new callback class that enables routing both the locally recorded and playout audio data to external module. A factory method that creates an AudioDeviceModule instance that also registers the AudioTransportCapture with the device is also added. BUG=webrtc:7337 R=solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/2753453002 . Cr-Commit-Position: refs/heads/master@{#17570}
2017-04-06 10:17:01 -07:00
"audio_device_data_observer.cc",
Remove unnecessary dependencies for voe when building with include_internal_audio_device==0. In particular and practical terms, this avoids pulling in AudioDeviceModuleImpl and associated classes, in Chrome. BUG= R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49999004 Cr-Commit-Position: refs/heads/master@{#9229}
2015-05-20 09:44:38 +02:00
"audio_device_impl.cc",
"audio_device_impl.h",
Adding AudioDeviceDataObserver interface Added new callback class that enables routing both the locally recorded and playout audio data to external module. A factory method that creates an AudioDeviceModule instance that also registers the AudioTransportCapture with the device is also added. BUG=webrtc:7337 R=solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/2753453002 . Cr-Commit-Position: refs/heads/master@{#17570}
2017-04-06 10:17:01 -07:00
"include/audio_device_data_observer.h",
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
]
Roll chromium_revision 2a70cb1..4662d4f (367468:368042) I had to fix the audio_device BUILD.gn which was forgotten back in https://codereview.webrtc.org/1536923003. It also contained a few missing source files and one library. Change log: https://chromium.googlesource.com/chromium/src/+log/2a70cb1..4662d4f Full diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f Changed dependencies: * src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/6d0c448..0f8e6e4 * src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/8a7662a..ebfcc9a DEPS diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f/DEPS No update to Clang. TBR=henrika@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1565093002 Cr-Commit-Position: refs/heads/master@{#11162}
2016-01-07 02:54:22 -08:00
if (is_android) {
sources += [
Add missing audio_device .h files. Bug: webrtc:7650 Change-Id: Ifbfcac49da9f761f666c6ff22b934e3a3ab783fd Reviewed-on: https://webrtc-review.googlesource.com/37003 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21482}
2018-01-02 15:00:11 +01:00
"android/audio_common.h",
Roll chromium_revision 2a70cb1..4662d4f (367468:368042) I had to fix the audio_device BUILD.gn which was forgotten back in https://codereview.webrtc.org/1536923003. It also contained a few missing source files and one library. Change log: https://chromium.googlesource.com/chromium/src/+log/2a70cb1..4662d4f Full diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f Changed dependencies: * src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/6d0c448..0f8e6e4 * src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/8a7662a..ebfcc9a DEPS diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f/DEPS No update to Clang. TBR=henrika@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1565093002 Cr-Commit-Position: refs/heads/master@{#11162}
2016-01-07 02:54:22 -08:00
"android/audio_device_template.h",
"android/audio_manager.cc",
"android/audio_manager.h",
"android/audio_record_jni.cc",
"android/audio_record_jni.h",
"android/audio_track_jni.cc",
"android/audio_track_jni.h",
"android/build_info.cc",
"android/build_info.h",
"android/opensles_common.cc",
"android/opensles_common.h",
"android/opensles_player.cc",
"android/opensles_player.h",
Adds support for OpenSL ES based audio capture on Android. NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that OpenSL ES is not accidentally activated in existing clients. There are still some unresolved issues to sort out before it can be utilized. Enables possibility to use OpenSL ES based audio in both directions for WebRTC. All unit tests and demo clients have been tested with the new implementation but the new support is behind a flag (see above). More testing is needed before it can be used in the field and additional support for hardware effects is still missing. BUG=webrtc:5925 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/2119633004 . Cr-Commit-Position: refs/heads/master@{#14290}
2016-09-19 15:44:09 +02:00
"android/opensles_recorder.cc",
"android/opensles_recorder.h",
Roll chromium_revision 2a70cb1..4662d4f (367468:368042) I had to fix the audio_device BUILD.gn which was forgotten back in https://codereview.webrtc.org/1536923003. It also contained a few missing source files and one library. Change log: https://chromium.googlesource.com/chromium/src/+log/2a70cb1..4662d4f Full diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f Changed dependencies: * src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/6d0c448..0f8e6e4 * src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/8a7662a..ebfcc9a DEPS diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f/DEPS No update to Clang. TBR=henrika@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1565093002 Cr-Commit-Position: refs/heads/master@{#11162}
2016-01-07 02:54:22 -08:00
]
libs = [
"log",
"OpenSLES",
]
Add support of AAudio in native WebRTC on Android O and above Bug: webrtc:8914 Change-Id: I016dd8fcebba1644c0a83e5f1460520545d4cdde Reviewed-on: https://webrtc-review.googlesource.com/56180 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22467}
2018-03-16 10:09:49 +01:00
if (rtc_enable_android_aaudio) {
sources += [
"android/aaudio_player.cc",
"android/aaudio_player.h",
"android/aaudio_recorder.cc",
"android/aaudio_recorder.h",
"android/aaudio_wrapper.cc",
"android/aaudio_wrapper.h",
]
libs += [ "aaudio" ]
}
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
if (build_with_mozilla) {
include_dirs += [
"/config/external/nspr",
"/nsprpub/lib/ds",
"/nsprpub/pr/include",
]
}
Roll chromium_revision 2a70cb1..4662d4f (367468:368042) I had to fix the audio_device BUILD.gn which was forgotten back in https://codereview.webrtc.org/1536923003. It also contained a few missing source files and one library. Change log: https://chromium.googlesource.com/chromium/src/+log/2a70cb1..4662d4f Full diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f Changed dependencies: * src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/6d0c448..0f8e6e4 * src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/8a7662a..ebfcc9a DEPS diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f/DEPS No update to Clang. TBR=henrika@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1565093002 Cr-Commit-Position: refs/heads/master@{#11162}
2016-01-07 02:54:22 -08:00
}
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
if (rtc_use_dummy_audio_file_devices) {
defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
} else {
if (is_linux) {
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
sources += [
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
"linux/alsasymboltable_linux.cc",
"linux/alsasymboltable_linux.h",
"linux/audio_device_alsa_linux.cc",
"linux/audio_device_alsa_linux.h",
"linux/audio_mixer_manager_alsa_linux.cc",
"linux/audio_mixer_manager_alsa_linux.h",
"linux/latebindingsymboltable_linux.cc",
"linux/latebindingsymboltable_linux.h",
]
Prefix ALSA and PULSE macros with WEBRTC_. This CL renames the internal macros LINUX_ALSA and LINUX_PULSE and adds the prefix WEBRTC_. Since these macros are internal to WebRTC, it is better to use a prefix. Bug: None Change-Id: I2a07fa569a4da168006cc36f32e4dbb98a75814b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171514 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30885}
2020-03-24 12:56:47 +01:00
defines += [ "WEBRTC_ENABLE_LINUX_ALSA" ]
Have audio_device only depend on X11 if use_x11 is set. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/2909343002 Cr-Commit-Position: refs/heads/master@{#18335}
2017-05-30 12:06:04 -07:00
libs = [ "dl" ]
Make building with X11 libraries optional. Desktop capturing on Linux will be disabled in this case, but everything can be built without any X11 development libraries installed. BUG=webrtc:5716,webrtc:8319 Change-Id: I01bd6a4b02816b407be19476e22ff073d264b496 Reviewed-on: https://webrtc-review.googlesource.com/32360 Reviewed-by: Henrik Andreassson (OOO until Jan 2) <henrika@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Joachim Bauch <jbauch@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21462}
2017-12-20 21:25:47 +01:00
if (rtc_use_x11) {
Have audio_device only depend on X11 if use_x11 is set. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/2909343002 Cr-Commit-Position: refs/heads/master@{#18335}
2017-05-30 12:06:04 -07:00
libs += [ "X11" ]
Make building with X11 libraries optional. Desktop capturing on Linux will be disabled in this case, but everything can be built without any X11 development libraries installed. BUG=webrtc:5716,webrtc:8319 Change-Id: I01bd6a4b02816b407be19476e22ff073d264b496 Reviewed-on: https://webrtc-review.googlesource.com/32360 Reviewed-by: Henrik Andreassson (OOO until Jan 2) <henrika@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Joachim Bauch <jbauch@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21462}
2017-12-20 21:25:47 +01:00
defines += [ "WEBRTC_USE_X11" ]
Have audio_device only depend on X11 if use_x11 is set. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/2909343002 Cr-Commit-Position: refs/heads/master@{#18335}
2017-05-30 12:06:04 -07:00
}
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
if (rtc_include_pulse_audio) {
Prefix ALSA and PULSE macros with WEBRTC_. This CL renames the internal macros LINUX_ALSA and LINUX_PULSE and adds the prefix WEBRTC_. Since these macros are internal to WebRTC, it is better to use a prefix. Bug: None Change-Id: I2a07fa569a4da168006cc36f32e4dbb98a75814b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171514 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30885}
2020-03-24 12:56:47 +01:00
defines += [ "WEBRTC_ENABLE_LINUX_PULSE" ]
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
}
Compile ADM pulse code by default. Pulse related code should still be disabled unless WEBRTC_ENABLE_LINUX_PULSE is defined but it will always be compiled. Bug: None Change-Id: If8a03aae445a8c73c3c347e275c5996368fe3088 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171513 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30886}
2020-03-25 16:38:01 +01:00
sources += [
"linux/audio_device_pulse_linux.cc",
"linux/audio_device_pulse_linux.h",
"linux/audio_mixer_manager_pulse_linux.cc",
"linux/audio_mixer_manager_pulse_linux.h",
"linux/pulseaudiosymboltable_linux.cc",
"linux/pulseaudiosymboltable_linux.h",
]
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
}
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
if (is_mac) {
sources += [
"mac/audio_device_mac.cc",
"mac/audio_device_mac.h",
"mac/audio_mixer_manager_mac.cc",
"mac/audio_mixer_manager_mac.h",
]
Factor framework dependencies out of audio_device_impl Bug: None Change-Id: I7d8d737134bb1a9dcf376cd39e74e73a5a6a0e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147723 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28733}
2019-08-01 12:19:31 +02:00
deps += [
":audio_device_impl_frameworks",
"../third_party/portaudio:mac_portaudio",
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
]
}
if (is_win) {
sources += [
"win/audio_device_core_win.cc",
"win/audio_device_core_win.h",
]
libs = [
# Required for the built-in WASAPI AEC.
"dmoguids.lib",
"wmcodecdspuuid.lib",
"amstrmid.lib",
"msdmo.lib",
Add a missing Windows library "oleaut32.lib" is required for VariantInit: https://docs.microsoft.com/en-us/windows/win32/api/oleauto/nf-oleauto-variantinit Bug: webrtc:11807 Change-Id: If0511571340e14407ad9402636a4a64d328fabca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180440 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Eric Astor <epastor@google.com> Cr-Commit-Position: refs/heads/master@{#31806}
2020-07-28 17:55:21 -04:00
"oleaut32.lib",
Add a gyp/gn option to use dummy audio file devices. Conceptually, dummy audio file devices are a "platform", like win/mac/linux, and so the conditional slots under include_internal_audio_device. When enabled, use_dummy_audio_file_devices disables whatever platform-specific audio layer would have been used and turns on dummy file device support. BUG= Review-Url: https://codereview.webrtc.org/2250483002 Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-16 18:21:18 -07:00
]
}
GN: Fix windows clang errors. Attempt 2. BUG=webrtc:6255 NOTRY=True Review-Url: https://codereview.webrtc.org/2281513002 Cr-Commit-Position: refs/heads/master@{#13942}
2016-08-26 13:31:24 -07:00
configs += [ ":audio_device_warnings_config" ]
GN: Audio device module 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 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 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=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/14259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
}
} else {
defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
}
if (!build_with_chromium) {
sources += [
# Do not link these into Chrome since they contain static data.
"dummy/file_audio_device_factory.cc",
"dummy/file_audio_device_factory.h",
]
}
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
}
Add gn target for audio_device_tests. Note (for myself) that this depends on https://codereview.webrtc.org/2219653004/ and https://codereview.webrtc.org/2214003002/, it should not be landed before them. NOTRY=True BUG=webrtc:6170,webrtc:5949 Review-Url: https://codereview.webrtc.org/2216423002 Cr-Commit-Position: refs/heads/master@{#13773}
2016-08-16 02:17:44 -07:00
Factor framework dependencies out of audio_device_impl Bug: None Change-Id: I7d8d737134bb1a9dcf376cd39e74e73a5a6a0e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147723 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28733}
2019-08-01 12:19:31 +02:00
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
visibility = [ ":*" ]
Convert GN libs lists to frameworks GN recently added support for Apple frameworks to link, rather than overloading the libs lists. This pulls .frameworks out of the libs lists, so that GN can stop supporting .frameworks in libs in the future. Bug: chromium:1052560 Change-Id: I263230ddd3c468061584423bba9e1f887503bcaa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178601 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Sylvain Defresne <sdefresne@chromium.org> Cr-Commit-Position: refs/heads/master@{#31632}
2020-07-03 10:19:30 +02:00
frameworks = [
Factor framework dependencies out of audio_device_impl Bug: None Change-Id: I7d8d737134bb1a9dcf376cd39e74e73a5a6a0e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147723 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28733}
2019-08-01 12:19:31 +02:00
# Needed for CoreGraphics:
"ApplicationServices.framework",
"AudioToolbox.framework",
"CoreAudio.framework",
# Needed for CGEventSourceKeyState in audio_device_mac.cc:
"CoreGraphics.framework",
]
}
}
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
rtc_source_set("mock_audio_device") {
testonly = true
sources = [
"include/mock_audio_device.h",
"include/mock_audio_transport.h",
Add missing audio_device .h files. Bug: webrtc:7650 Change-Id: Ifbfcac49da9f761f666c6ff22b934e3a3ab783fd Reviewed-on: https://webrtc-review.googlesource.com/37003 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21482}
2018-01-02 15:00:11 +01:00
"mock_audio_device_buffer.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
deps = [
":audio_device",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
":audio_device_buffer",
":audio_device_impl",
Replace MockAudioDeviceModule mock refcounting with real refcounting Bug: webrtc:11308 Change-Id: Ic55ec2c4b45f8fc709fe1348556bdeea6202e7a8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166580 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30366}
2020-01-22 15:23:43 -08:00
"../../rtc_base:refcount",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../test:test_support",
]
}
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -08:00
if (rtc_include_tests) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_device_unittests") {
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -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
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -08:00
sources = [
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
"fine_audio_buffer_unittest.cc",
Cleanup after moving test/fake_audio_device. Cleanup after moving test/fake_audio_device to modules/audio_device/include/test_audio_device. Hide implementation of test audio device module in the anonymous namespace. Bug: webrtc:8946 Change-Id: I2d49c3ec5d43eeb5f155d38de95f69ed3c537805 Reviewed-on: https://webrtc-review.googlesource.com/61426 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22401}
2018-03-13 13:25:19 +01:00
"include/test_audio_device_unittest.cc",
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -08:00
]
deps = [
":audio_device",
Split up audio_device build target We currently have one build target containing everything for audio_device: the interfaces, the "fine" audio buffer, and the actual implementations for each platform. Since we are planning to move the Android implementation to the sdk/android folder, we only want to depend on the interfaces and the "fine" audio buffer, not the other platform specific implementations. This CL splits the audio_device target into three different targets: the interfaces, the fine audio buffer, and the platform specific implementations. The default audio_device target now points to the interfaces instead. Bug: webrtc:7452 Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd Reviewed-on: https://webrtc-review.googlesource.com/61321 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22452}
2018-03-15 12:44:12 +01:00
":audio_device_buffer",
":audio_device_impl",
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
":mock_audio_device",
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",
Propagate TaskQueueFactory to AudioDeviceBuffer keep using GlobalTaskQueueFactory in android/ios bindings. Switch to DefaultTaskQueueFactory in tests. Bug: webrtc:10284 Change-Id: I034c70542be5eeb830be86527830d51204fb2855 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 09:16:12 +02:00
"../../api/task_queue",
"../../api/task_queue:default_task_queue_factory",
Separate test/fake_audio_device on API and implementation. Step 1. Adding ability of injecting audio in end to end tests, that are using WebRTC. It will be done in 3 steps: 1. Test/fake_audio_device will be moved to production part of WebRTC source code and renamed to test_audio_device_module. Old header is replaced with alias to the new one. 2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice. 3. test/fake_audio_device will be removed. This CL implements 1st step. Bug: webrtc:8946 Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f Reviewed-on: https://webrtc-review.googlesource.com/59740 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22325}
2018-03-07 12:20:51 +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:rtc_base_approved",
Migrate modules/audio_device to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I6d1a7145aaaae2e4cd0c8658fa31a673f857dbd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178814 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31664}
2020-07-08 10:13:20 +02:00
"../../rtc_base/synchronization:mutex",
Decoupling audio_device from Obj-C code The goal of this CL is to separate Obj-C/Obj-C++ code from targets which have also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743 for more information). To achieve this we have created 2 targets (audio_device_ios_objc and audio_device_generic) and audio_device will act as a proxy between these targets (this way we can avoid a circular dependency between audio_device_generic and audio_device_ios_objc). BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2991343002 Cr-Commit-Position: refs/heads/master@{#19795}
2017-09-12 04:45:24 -07:00
"../../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",
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -08:00
"../../test:test_support",
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
"../utility",
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -08:00
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
ADM unit test now runs on Mac and Windows as well. TBR=solenberg BUG=webrtc:7273 Review-Url: https://codereview.webrtc.org/2783863002 Cr-Commit-Position: refs/heads/master@{#17451}
2017-03-29 07:50:19 -07:00
if (is_linux || is_mac || is_win) {
Adds unit test for ADM on Linux BUG=webrtc:7273 Review-Url: https://codereview.webrtc.org/2736503002 Cr-Commit-Position: refs/heads/master@{#17285}
2017-03-17 04:26:22 -07:00
sources += [ "audio_device_unittest.cc" ]
}
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
if (is_win) {
Ports CoreAudioUtil from Chrome to WebRTC. See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34 for details. Bug: webrtc:9265 Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb Reviewed-on: https://webrtc-review.googlesource.com/76900 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:20:58 +02:00
sources += [ "win/core_audio_utility_win_unittest.cc" ]
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
deps += [
":audio_device_module_from_input_and_output",
":windows_core_audio_utility",
]
Ports CoreAudioUtil from Chrome to WebRTC. See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34 for details. Bug: webrtc:9265 Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb Reviewed-on: https://webrtc-review.googlesource.com/76900 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:20:58 +02:00
}
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
if (is_android) {
# Need to disable error due to the line in
# base/android/jni_android.h triggering it:
# const BASE_EXPORT jobject GetApplicationContext()
# error: type qualifiers ignored on function return type
cflags = [ "-Wno-ignored-qualifiers" ]
sources += [
"android/audio_device_unittest.cc",
"android/audio_manager_unittest.cc",
"android/ensure_initialized.cc",
"android/ensure_initialized.h",
]
deps += [
Fix paths that were trying to go above the root by using too many "../.." This doesn't cause problems because seems like GN just ignores these attempts and confines everything to the root. Bug: None Change-Id: Ief50b6e10f3950b319d0bde547f91dcc7f237cda Reviewed-on: https://webrtc-review.googlesource.com/90041 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24063}
2018-07-23 10:04:12 +02:00
"../../base",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../../sdk/android:libjingle_peerconnection_java",
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
]
}
Fix linker error when building without internal audio device. Bug: webrtc:8678 Change-Id: I30f0d1aa7bec6931edb9e5ba00a6813dea1536ff Reviewed-on: https://webrtc-review.googlesource.com/35480 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21408}
2017-12-20 21:19:49 +01:00
if (!rtc_include_internal_audio_device) {
defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
}
Created a mocked AudioTransport. There are currently two nearly identical classes called MockAudioTransport defined in two unit tests: android/audio_transport_unittest.cc and /ios/audio_transport_unittest_ios.cc This change defines a common mocked AudioTransport. The two current mocks are rewritten to use the common one. A GN target is created for this mock and MockAudioDevice. This change will allow to provide a mocked AudioTransport to AudioState in a dependent CL https://codereview.webrtc.org/2454373002/ BUG=webrtc:6346 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2493483002 Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 01:05:34 -08:00
}
Add gn target for audio_device_tests. Note (for myself) that this depends on https://codereview.webrtc.org/2219653004/ and https://codereview.webrtc.org/2214003002/, it should not be landed before them. NOTRY=True BUG=webrtc:6170,webrtc:5949 Review-Url: https://codereview.webrtc.org/2216423002 Cr-Commit-Position: refs/heads/master@{#13773}
2016-08-16 02:17:44 -07:00
}
Reland of Fixing package-boundary violation with srjar_deps (patchset #1 id:1 of https://codereview.webrtc.org/2617533005/ ) Reason for revert: Skipping the build of the target "//webrtc/modules/audio_device:audio_device_java" when building webrtc with chromium. This was causing a failure because in that case it is not possible to refer to the script used by the template "android_shared_srcjar" with an absolute path (which seems to be necessary since the script attribute of action is relative to the invocation target and not to the template declaration). Original issue's description: > Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ ) > > Reason for revert: > This CL is breaking a chromium.webrtc.fyi: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/226 > > I am trying to reproduce the issue on my local machine and I will try to re-land the CL later. > > Original issue's description: > > Fixing package-boundary violation with srjar_deps > > > > Without the usage of the srcjar_deps attribute we were not able to > > include .java files from other packages without violating the package > > boundary contraint. > > > > As an example, in this CL the target "libjingle_peerconnection_java" was > > directly including .java files from another packages in its "java_files" > > attribute. > > > > Using srcjar_deps we are able to declare the dependency of the target > > avoiding to create hidden dependencies in the codebase. > > > > This is not fixing the webrtc:6356 bug directly but it is a first step to > > include ThreadUtils classes in libjingle_peerconnection_client_java.jar > > again. > > > > It seems also to be related to the chromium:648244 bug. This can be solved > > if we can find a way to perform srcjar generation in the android_library > > target without changing the semantic of the target. > > > > BUG=webrtc:6356 > > > > Review-Url: https://codereview.webrtc.org/2610823002 > > Cr-Commit-Position: refs/heads/master@{#15914} > > Committed: https://chromium.googlesource.com/external/webrtc/+/10a76592a78762187bfaf3547b7e9986454062ab > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6356 > > Review-Url: https://codereview.webrtc.org/2617533005 > Cr-Commit-Position: refs/heads/master@{#15915} > Committed: https://chromium.googlesource.com/external/webrtc/+/eb731ed09ebfea1b26d1b7bc1f2678f7bbc012ee TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6356 Review-Url: https://codereview.webrtc.org/2612953004 Cr-Commit-Position: refs/heads/master@{#15958}
2017-01-09 02:37:21 -08:00
if (!build_with_chromium && is_android) {
Reland: "Make javac warnings errors for WebRTC targets." This reverts commit 2bad72a27329ff30ceb9479253f5eb3d21888d25. Reason for revert: Fixing downstream projects (take 2). Original change's description: > Reland "Revert "Make javac warnings errors for WebRTC targets."" > > This is a reland of 098d24c3c18f4b1fd043d7ba716d7601f0ce2b74 > Original change's description: > > Revert "Make javac warnings errors for WebRTC targets." > > > > This reverts commit 19b761403c3522902d69d61179f4d184e3632f79. > > > > Reason for revert: Breaking internal builds > > > > Original change's description: > > > Make javac warnings errors for WebRTC targets. > > > > > > Adds new rtc_* templates for Android targets to allow specifying > > > default values that affect WebRTC targets. > > > > > > Bug: webrtc:6597 > > > Change-Id: Ie529bfc8500d1e785b8a59dba7078b5f88ccfcd1 > > > Reviewed-on: https://webrtc-review.googlesource.com/15103 > > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > > > Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#20567} > > > TBR=phoglund@webrtc.org,sakal@webrtc.org > > > > Change-Id: I6d3ff5604b3d4307765d3a65adb783f89fcc974c > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:6597 > > Reviewed-on: https://webrtc-review.googlesource.com/20740 > > Reviewed-by: Lu Liu <lliuu@webrtc.org> > > Commit-Queue: Lu Liu <lliuu@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20571} > > Bug: webrtc:6597 > Change-Id: Icfb5ded46ce76b674bae67bfa02054b4ec52bb0f > Reviewed-on: https://webrtc-review.googlesource.com/20800 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20577} TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org,mbonadei@webrtc.org,sakal@webrtc.org,lliuu@webrtc.org Change-Id: Id3713c1885318741711987ae642a269a9ca5bb85 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:6597 Reviewed-on: https://webrtc-review.googlesource.com/18441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20588}
2017-11-07 15:36:33 +00:00
rtc_android_library("audio_device_java") {
[Android] Replace java_files with sources Replace all usages of java_files with sources in gn files, and automatically format. This is in preparation for java_files being completely removed upstream in favor of sources. NOPRESUBMIT=true Bug: chromium:1035074 Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Natalie Chouinard <chouinard@google.com> Cr-Commit-Position: refs/heads/master@{#30135}
2019-12-23 14:02:25 -08:00
sources = [
Reland of Fixing package-boundary violation with srjar_deps (patchset #1 id:1 of https://codereview.webrtc.org/2617533005/ ) Reason for revert: Skipping the build of the target "//webrtc/modules/audio_device:audio_device_java" when building webrtc with chromium. This was causing a failure because in that case it is not possible to refer to the script used by the template "android_shared_srcjar" with an absolute path (which seems to be necessary since the script attribute of action is relative to the invocation target and not to the template declaration). Original issue's description: > Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ ) > > Reason for revert: > This CL is breaking a chromium.webrtc.fyi: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/226 > > I am trying to reproduce the issue on my local machine and I will try to re-land the CL later. > > Original issue's description: > > Fixing package-boundary violation with srjar_deps > > > > Without the usage of the srcjar_deps attribute we were not able to > > include .java files from other packages without violating the package > > boundary contraint. > > > > As an example, in this CL the target "libjingle_peerconnection_java" was > > directly including .java files from another packages in its "java_files" > > attribute. > > > > Using srcjar_deps we are able to declare the dependency of the target > > avoiding to create hidden dependencies in the codebase. > > > > This is not fixing the webrtc:6356 bug directly but it is a first step to > > include ThreadUtils classes in libjingle_peerconnection_client_java.jar > > again. > > > > It seems also to be related to the chromium:648244 bug. This can be solved > > if we can find a way to perform srcjar generation in the android_library > > target without changing the semantic of the target. > > > > BUG=webrtc:6356 > > > > Review-Url: https://codereview.webrtc.org/2610823002 > > Cr-Commit-Position: refs/heads/master@{#15914} > > Committed: https://chromium.googlesource.com/external/webrtc/+/10a76592a78762187bfaf3547b7e9986454062ab > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6356 > > Review-Url: https://codereview.webrtc.org/2617533005 > Cr-Commit-Position: refs/heads/master@{#15915} > Committed: https://chromium.googlesource.com/external/webrtc/+/eb731ed09ebfea1b26d1b7bc1f2678f7bbc012ee TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6356 Review-Url: https://codereview.webrtc.org/2612953004 Cr-Commit-Position: refs/heads/master@{#15958}
2017-01-09 02:37:21 -08:00
"android/java/src/org/webrtc/voiceengine/BuildInfo.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
]
Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ ) Reason for revert: It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places. Original issue's description: > Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ ) > > Reason for revert: > This breaks some chromium.webrtc.fyi buildbots with the following error: > > ERROR Unresolved dependencies. > //third_party/webrtc/base:base(//build/toolchain/android:android_arm) > needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm) > > > Original issue's description: > > Creating libwebrtc bundle jar > > > > Creates a JAR which includes: > > - //webrtc/base:base_java > > - //webrtc/modules/audio_device:audio_device_java > > - //webrtc/sdk/android:libjingle_peerconnection_java > > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java > > > > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'. > > > > BUG=webrtc:6356 > > > > Review-Url: https://codereview.webrtc.org/2646443002 > > Cr-Commit-Position: refs/heads/master@{#16189} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a62a82b7e7da5a1bbbf8b5614ef19334cc1603ce > > TBR=kjellander@webrtc.org,sakal@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6356 > > Review-Url: https://codereview.webrtc.org/2640023010 > Cr-Commit-Position: refs/heads/master@{#16190} > Committed: https://chromium.googlesource.com/external/webrtc/+/3c9151b9532b372a35f0b81cc7c6b34d820506e1 TBR=kjellander@webrtc.org,sakal@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6356 Review-Url: https://codereview.webrtc.org/2646093004 Cr-Commit-Position: refs/heads/master@{#16203}
2017-01-23 01:25:53 -08:00
deps = [
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../../rtc_base:base_java",
Android: Use android_deps directly This is preparing for upstream removing the alias java_groups for the individual support library targets: https://crrev.com/c/1500780 Bug: chromium:937987 Change-Id: I1c9efd83f6997288b334f3dc2f41233fa4e2ab61 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Peter Wen <wnwen@chromium.org> Cr-Commit-Position: refs/heads/master@{#26995}
2019-03-06 09:18:57 -05:00
"//third_party/android_deps:com_android_support_support_annotations_java",
Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ ) Reason for revert: It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places. Original issue's description: > Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ ) > > Reason for revert: > This breaks some chromium.webrtc.fyi buildbots with the following error: > > ERROR Unresolved dependencies. > //third_party/webrtc/base:base(//build/toolchain/android:android_arm) > needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm) > > > Original issue's description: > > Creating libwebrtc bundle jar > > > > Creates a JAR which includes: > > - //webrtc/base:base_java > > - //webrtc/modules/audio_device:audio_device_java > > - //webrtc/sdk/android:libjingle_peerconnection_java > > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java > > > > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'. > > > > BUG=webrtc:6356 > > > > Review-Url: https://codereview.webrtc.org/2646443002 > > Cr-Commit-Position: refs/heads/master@{#16189} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a62a82b7e7da5a1bbbf8b5614ef19334cc1603ce > > TBR=kjellander@webrtc.org,sakal@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6356 > > Review-Url: https://codereview.webrtc.org/2640023010 > Cr-Commit-Position: refs/heads/master@{#16190} > Committed: https://chromium.googlesource.com/external/webrtc/+/3c9151b9532b372a35f0b81cc7c6b34d820506e1 TBR=kjellander@webrtc.org,sakal@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6356 Review-Url: https://codereview.webrtc.org/2646093004 Cr-Commit-Position: refs/heads/master@{#16203}
2017-01-23 01:25:53 -08:00
]
Reland of Fixing package-boundary violation with srjar_deps (patchset #1 id:1 of https://codereview.webrtc.org/2617533005/ ) Reason for revert: Skipping the build of the target "//webrtc/modules/audio_device:audio_device_java" when building webrtc with chromium. This was causing a failure because in that case it is not possible to refer to the script used by the template "android_shared_srcjar" with an absolute path (which seems to be necessary since the script attribute of action is relative to the invocation target and not to the template declaration). Original issue's description: > Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ ) > > Reason for revert: > This CL is breaking a chromium.webrtc.fyi: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/226 > > I am trying to reproduce the issue on my local machine and I will try to re-land the CL later. > > Original issue's description: > > Fixing package-boundary violation with srjar_deps > > > > Without the usage of the srcjar_deps attribute we were not able to > > include .java files from other packages without violating the package > > boundary contraint. > > > > As an example, in this CL the target "libjingle_peerconnection_java" was > > directly including .java files from another packages in its "java_files" > > attribute. > > > > Using srcjar_deps we are able to declare the dependency of the target > > avoiding to create hidden dependencies in the codebase. > > > > This is not fixing the webrtc:6356 bug directly but it is a first step to > > include ThreadUtils classes in libjingle_peerconnection_client_java.jar > > again. > > > > It seems also to be related to the chromium:648244 bug. This can be solved > > if we can find a way to perform srcjar generation in the android_library > > target without changing the semantic of the target. > > > > BUG=webrtc:6356 > > > > Review-Url: https://codereview.webrtc.org/2610823002 > > Cr-Commit-Position: refs/heads/master@{#15914} > > Committed: https://chromium.googlesource.com/external/webrtc/+/10a76592a78762187bfaf3547b7e9986454062ab > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6356 > > Review-Url: https://codereview.webrtc.org/2617533005 > Cr-Commit-Position: refs/heads/master@{#15915} > Committed: https://chromium.googlesource.com/external/webrtc/+/eb731ed09ebfea1b26d1b7bc1f2678f7bbc012ee TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6356 Review-Url: https://codereview.webrtc.org/2612953004 Cr-Commit-Position: refs/heads/master@{#15958}
2017-01-09 02:37:21 -08:00
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}
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