https://codereview.webrtc.org/2514883002/ changed and moved these targets around but did not add public dependencies for the fallbacks, which causes gn gen --check a lot of anger.
NOTRY=true # Only build changes and windows bots are cranky atm.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2651663002
Cr-Commit-Position: refs/heads/master@{#16214}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
The factory follows the same principles as PeerConnectionFactory;
various modules can be passed into its constructor but default
implementations are provided. Currently the only object the factory can
create is a UdpTransport (need to start somewhere).
UdpTransportChannel (renamed to UdpTransport)
will now accept a socket passed into its constructor,
relying on the factory to create the socket. This allows some
simplifications to be made, such as getting rid of "State" since the
only states are now "has destination set or doesn't".
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2632613002
Cr-Commit-Position: refs/heads/master@{#16154}
This maps, in both directions, [Audio/Video]TrackInterface with
[Voice/Video][Sender/Receiver]Info.
This mapping is necessary for RTCStatsCollector to know the relationship
between RTCMediaStreamTrackStats and RTC[In/Out]boundRTPStreamStats, and
to be able to collect several RTCMediaStreamTrackStats stats.
BUG=webrtc:6757, chromium:659137, chromium:657854, chromium:627816
Review-Url: https://codereview.webrtc.org/2611983002
Cr-Commit-Position: refs/heads/master@{#16090}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
This allows building without SCTP support (and even building/running
tests). The "HAVE_SCTP" define has been functional for a while, but there
wasn't any easy way to turn it on/off.
NOTRY=True
BUG=webrtc:6933
Review-Url: https://codereview.webrtc.org/2593313002
Cr-Commit-Position: refs/heads/master@{#15763}
Also rename internal::FlexfecReceiveStream to FlexfecReceiveStreamImpl.
BUG=webrtc:6849
Review-Url: https://codereview.webrtc.org/2561123002
Cr-Commit-Position: refs/heads/master@{#15666}
This moves some GN check configurations out of .gn to individual targets.
The now checked targets are:
"//webrtc/api/*",
"//webrtc/audio/*",
"//webrtc/modules/audio_coding/*",
Many targets were fixed by adding dependencies, but the ones that
requires more refactorings are left with the check_includes attribute
set to false instead.
Make //webrtc/test:test_support a public dep of //webrtc/test:test_main
to avoid having to add that to all users of it.
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2556943003
Cr-Commit-Position: refs/heads/master@{#15461}
I decided to make one webrtc/sdk/android/BUILD.gn for both tests and Java/jni src.
External dependencies needs to be updated after this CL.
Future work is required to clean up the Android api and move
implementation details to /webrtc/sdk/android/src.
BUG=webrtc:5882,webrtc:6804
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2547483003
Cr-Commit-Position: refs/heads/master@{#15443}
This is in preparation for https://codereview.webrtc.org/2517173004/,
which needs some updates of downstream dependencies. This cl adds the
target to api/BUILD.gn, creates the directory api/video, and a single
harmless include file there.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2546723003
Cr-Commit-Position: refs/heads/master@{#15385}
This is an integration test using peerconnectiontestwrapper.h to set up
and end to end test using a real PeerConnection implementation. These
tests will complement rtcstatscollector_unittest.cc which collects all
stats using mocks.
The integration test is set up so that all stats types are returned by
GetStats and verifies that expected dictionary members are defined. The
test could in the future be updated to include sanity checks for the
values of members. There is a sanity check that references to other
stats dictionaries yield existing stats of the appropriate type, but
other than that members are only tested for if they are defined not.
StatsCallback of rtcstatscollector_unittest.cc is moved so that it can
be reused and renamed to RTCStatsObtainer.
TODO: Audio stream track stats members are missing in the test. Find out
if this is because of a real problem or because of testing without real
devices. Do this before closing crbug.com/627816.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2521663002
Cr-Commit-Position: refs/heads/master@{#15287}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
The mock is used in a dependent CL https://codereview.webrtc.org/2436033002.
There is also a goal to allow external mixing implementations
(subclasses of webrtc::AudioMixer) and inject them to
PeerConnectionFactory. We think that part of that is an official and
maintained mock.
Summary of changes:
* Created a mixer mock/stub in webrtc/api/test
* Made a target webrtc/api:mock_audio_mixer for it.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2520323002
Cr-Commit-Position: refs/heads/master@{#15190}
This file fits there more naturally since it has dependencies to jni.
BUG=None
Review-Url: https://codereview.webrtc.org/2514383002
Cr-Commit-Position: refs/heads/master@{#15179}
This allows downstream dependencies can add it as a dependency.
BUG=webrtc:6499
Review-Url: https://codereview.webrtc.org/2521183002
Cr-Commit-Position: refs/heads/master@{#15178}
The peer connection loopback test could catch regressions too, but it's
too slow to run on downstream ARM emulators. I'm adding a test here
that just makes sure we can load the JNI and init audio/video engines
in WebRTC.
This test overlaps in functionality with the existing tests,
but we need it anyway since all existing tests are too timing-sensitive.
Removes resources from the test; they're awkward downstream and we
don't really need them anyway.
BUG=b/32820229
Review-Url: https://codereview.webrtc.org/2506603002
Cr-Commit-Position: refs/heads/master@{#15101}
The audio_device_module field was currently unused. The audio_mixer
field is going to be used to pass an AudioMixer to AudioState.
In the hopefully-not-very-far future, the toplevel WebRTC API will allow passing
a custom AudioMixer, e.g. for spatialized audio (audio in space). If no
mixer is passed, a default mixer is created (the one in modules/audio_mixer).
The only object which will have a permanent reference to the mixer is AudioState.
AudioState is created in WebRTCVoiceEngine with a configuration object,
which already contains a VoiceEngine pointer. In this CL, we extend this
config object with a mixer pointer.
In summary: in an upcoming CL, a mixer will be either created in or passed to
WebRTCVoiceEngine. This mixer will be passed to the ctor of AudioState in a
config struct.
BUG=webrtc:6346
NOTRY=True
Review-Url: https://codereview.webrtc.org/2456363002
Cr-Commit-Position: refs/heads/master@{#14973}
As I was preparing to move some files from the api/ folder, I noticed
that this file was not included in the BUILD.gn file. I've added it back
in and updated it to compile and run successfully again.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2485603002
Cr-Commit-Position: refs/heads/master@{#14965}
Essentially applying the same change as in
https://codereview.webrtc.org/2023413002 in more locations.
There's only one change affecting production code: enabling the warning
for webrtc/media:rtc_media. The rest are test changes.
With these changes, the only place the warning is disabled is in the Windows
implementation of webrtc/modules/video_capture:video_capture_internal_impl,
which is harder to fix, since it relies on sample code from the Windows SDK.
BUG=webrtc:6653
NOTRY=True
Review-Url: https://codereview.webrtc.org/2468093004
Cr-Commit-Position: refs/heads/master@{#14938}
Introduce a convention on categorizing GN targets:
1. Production code
2. Tests
3. Examples
4. Tools
The first two have targets spread out all over the tree,
while the latter are isolated to examples/ and tools/ directories.
Another new convention: Each directory's BUILD.gn file shall contain
a target named similar to the directory name. This target shall
contain the 'most common' production code, i.e. so that most
consumers of the directory can depend on only the directory
(which implicitly means that target in GN).
//webrtc:webrtc_tests is changed to depend on all WebRTC tests.
From now on, it's necessary to add new test targets to this dependency
tree in order to get them compiled.
Two new group targets are created:
//webrtc/modules/audio_coding:audio_coding_tests
//webrtc/modules/audio_processing:audio_processing_tests
to reduce the long list of tests in //webrtc:webrtc_tests.
Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted
to the root target, to avoid circular dependencies due to the monolithic
property of these targets (a problem we've had in the past).
The 'root' target at the top level is renamed to 'default', which means GN will
build this target instead of _all_ generated targets
(see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default).
This target now depends on everything we want to build, meaning all targets now
explicitly needs to be wired up from the root target in order to get build.
Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build.
BUG=webrtc:6440
TESTED=git cl try --clobber
NOTRY=True
Review-Url: https://codereview.webrtc.org/2441383002
Cr-Commit-Position: refs/heads/master@{#14821}
As a side effect:
- Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
- Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
- Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
BUG=webrtc:4690
Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Original-Commit-Position: refs/heads/master@{#14771}
Cr-Commit-Position: refs/heads/master@{#14780}
Reason for revert:
Breaks downstream project
Original issue's description:
> Clean up logging in AudioSendStream::SetupSendCodec().
>
> As a side effect:
> - Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
> - Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
> - Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
> Cr-Commit-Position: refs/heads/master@{#14771}
TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2452643002
Cr-Commit-Position: refs/heads/master@{#14774}
As a side effect:
- Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
- Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
- Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Commit-Position: refs/heads/master@{#14771}
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.
This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.
It will also create less build dependencies when the new mixer has replaced the old one.
NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
This class is logically parallel with the {Audio,Video}ReceiveStream
classes. Its purpose is to describe a receive stream of FlexFEC packets,
through the corresponding config.
Functionally, this class simply forwards the received RTP packets
to its FlexfecReceiver, which returns recovered packets to the
Call level, for appropriate demultiplexing based on SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2397843005
Cr-Commit-Position: refs/heads/master@{#14704}
YuvConverter is complex class that deserves its own file. It is also used outside of SurfaceTextureHelper.
BUG=webrtc:6470
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2426023002 .
Cr-Commit-Position: refs/heads/master@{#14683}
This makes it possible for external applications to use this class.
BUG=webrtc:6524
NOTRY=True
Review-Url: https://codereview.webrtc.org/2430693002
Cr-Commit-Position: refs/heads/master@{#14679}
Reason for revert:
Breaks internal project.
Original issue's description:
> Support for video file instead of camera and output video out to file
>
> When video out to file is enabled the remote video which is recorded is
> not show on screen.
>
> You can use this command line for file input and output:
> monkeyrunner ./webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py --devname 02157df28cd47001 --videoin /storage/emulated/0/reference_video_1280x720_30fps.y4m --videoout /storage/emulated/0/output.y4m --videoout_width 1280 --videoout_height 720 --videooutsave /tmp/out.y4m
>
> BUG=webrtc:6545
>
> Committed: https://crrev.com/44666997ca912705f8f96c9bd211e719525a3ccc
> Cr-Commit-Position: refs/heads/master@{#14660}
TBR=magjed@webrtc.org,sakal@webrtc.org,jansson@chromium.org,mandermo@google.com,mandermo@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2425763003
Cr-Commit-Position: refs/heads/master@{#14664}
When video out to file is enabled the remote video which is recorded is
not show on screen.
You can use this command line for file input and output:
monkeyrunner ./webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py --devname 02157df28cd47001 --videoin /storage/emulated/0/reference_video_1280x720_30fps.y4m --videoout /storage/emulated/0/output.y4m --videoout_width 1280 --videoout_height 720 --videooutsave /tmp/out.y4m
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2273573003
Cr-Commit-Position: refs/heads/master@{#14660}
The purpose is to prepare for a TextureViewRenderer that will share the
EGL rendering code.
Two functional changes are also included:
* The implementation of SurfaceHolder.Callback.surfaceDestroyed will now
block until the EGL surface is released. This is done in order to
comply with the documentation that says: "If you have a rendering
thread that directly accesses the surface, you must ensure that thread
is no longer touching the Surface before returning from this function."
* We will no longer try to hide render glitches during layout changes.
This was a lost cause anyway.
BUG=webrtc:6407
Review-Url: https://codereview.webrtc.org/2399463006
Cr-Commit-Position: refs/heads/master@{#14570}
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}
New file structure and targets:
rtc_stats_api
webrtc/api/stats/rtcstats.h
webrtc/api/stats/rtcstats_objects.h
webrtc/api/stats/rtcstatsreport.h
rtc_stats (dep on rtc_stats_api)
webrtc/stats/rtcstats.cc
webrtc/stats/rtcstats_objects.cc
webrtc/stats/rtcstatsreport.cc
libjingle_peerconnection (dep on rtc_stats)
webrtc/api/rtcstatscollector.cc
webrtc/api/rtcstatscollector.h
Placing rtc_stats_api headers in this separate target instead of
libjingle_peerconnection avoids a circular dependency
libjingle_peerconnection -> rtc_stats -> libjingle_peerconnection
Code changes:
PeerConnectionInterface::GetStats(RTCStatsCollectorCallback*) added for
the new stats collection API. Implemented by PeerConnection.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2331373004
Cr-Commit-Position: refs/heads/master@{#14246}
During GN vs GYP auditing it was discovered that some
GN targets that had public_configs were not exposing them
to dependents where the dependent depended on a group, which
in turn included that target as a dependency. Instead of
changing those public_configs to all_dependent_configs
(which would be a change from GYP), it's better to just change
those group targets to use public_deps instead.
BUG=webrtc:6323
NOTRY=True
TESTED=Generated GYP and GN project files on Mac and ran the
tools/gyp_flag_compare.py script before and after this patch was
applied. The file in question used for inspection was the
webrtc/api/webrtcsessiondescriptionfactory.cc
which is a part of the libjingle_peerconnection target.
Review-Url: https://codereview.webrtc.org/2344623002
Cr-Commit-Position: refs/heads/master@{#14222}
