This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
- Out from modules/utility/ and into modules/audio_device/ios/ - there they are used.
BUG=none
Review-Url: https://codereview.webrtc.org/2526273002
Cr-Commit-Position: refs/heads/master@{#15236}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
They are implementations of interfaces that are only ever exposed
via "create" functions, so the entire class definitions can be put in
anonymous namespaces in the .cc files that defines the "create"
functions.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2038513002
Cr-Commit-Position: refs/heads/master@{#13794}
Reason for revert:
Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now.
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This has been landed twice before, as
> https://codereview.webrtc.org/2037623002 and
> https://codereview.webrtc.org/2240163002. Third time's a charm!)
>
> NOPRESUBMIT=True
> TBR=kjellander@webrtc.org
>
> Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef
> Cr-Commit-Position: refs/heads/master@{#13777}
TBR=
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2245413002
Cr-Commit-Position: refs/heads/master@{#13779}
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).
Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.
This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163eTBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
These parts were commented out to avoid breaking the Chromium
WebRTC FYI bots. Include them in the WebRTC build to make our bots
build as many as possible of our GN targets.
BUG=webrtc:5949
NOTRY=True
TBR=phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/2054903002
Cr-Commit-Position: refs/heads/master@{#13097}
Reason for revert:
Reland. It was a false alarm.
Original issue's description:
> Revert of Change ProcessThread's task type to be the one from TaskQueue. (patchset #3 id:80001 of https://codereview.webrtc.org/2016043003/ )
>
> Reason for revert:
> Downstream issues
>
> Original issue's description:
> > Change ProcessThread's task type to be the one from TaskQueue.
> > ProcessThread will eventually be replaced by TaskQueue, so this is the first little step.
> >
> > BUG=
> > R=magjed@webrtc.org
> >
> > Committed: https://crrev.com/400a276c8a0b299190ff17a81edd8780a26d63d3
> > Cr-Commit-Position: refs/heads/master@{#12952}
>
> TBR=magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=
>
> Committed: https://crrev.com/641455176a1241dea6dda7071ba4162f41a0b5fc
> Cr-Commit-Position: refs/heads/master@{#12953}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2017333002
Cr-Commit-Position: refs/heads/master@{#12954}
Reason for revert:
Downstream issues
Original issue's description:
> Change ProcessThread's task type to be the one from TaskQueue.
> ProcessThread will eventually be replaced by TaskQueue, so this is the first little step.
>
> BUG=
> R=magjed@webrtc.org
>
> Committed: https://crrev.com/400a276c8a0b299190ff17a81edd8780a26d63d3
> Cr-Commit-Position: refs/heads/master@{#12952}
TBR=magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2020783003
Cr-Commit-Position: refs/heads/master@{#12953}
Compile fixes for GN on iOS that finally gets our bots green.
Changes to system_wrappers:
* Updated to only use inclusive sources for maintainability
* Add a few missing GN headers.
* Cleanup GYP hack for atomic32_mac.cc
* Renamed changes sources to avoid problems with GYP/GN file
suffix rules:
- atomic32_mac.cc -> atomic32_darwin.cc
- atomic32_posix.cc -> atomic32_non_darwin_unix.cc
See https://code.google.com/p/chromium/codesearch#chromium/src/build/config/BUILDCONFIG.gn&l=325
for details on which extensions can/cannot be used.
BUG=webrtc:5586
NOTRY=True
Review-Url: https://codereview.webrtc.org/1999723002
Cr-Commit-Position: refs/heads/master@{#12897}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.
The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.
BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
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
