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}
libjingle_peerconnection_so is not including common_config, which is
causing some differences is the defines.
We'd like to prevent that happening in the future.
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2325603002
Cr-Commit-Position: refs/heads/master@{#14127}
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}
Reason for revert:
Downstream apps should now be fixed.
Original issue's description:
> Revert of Remove the old AndroidVideoCapturer stack code. (patchset #2 id:20001 of https://codereview.webrtc.org/2235893003/ )
>
> Reason for revert:
> Breaks downstream.
>
> Original issue's description:
> > Remove the old AndroidVideoCapturer stack code.
> >
> > This code is no longer needed. Apps should be using the new API introduced here: https://codereview.webrtc.org/2127893002/
> >
> > Committed: https://crrev.com/1b365a8db070f9cdcbf35ec871f758dcd909e51d
> > Cr-Commit-Position: refs/heads/master@{#13950}
>
> TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/e39f251dacf66e50153bcda615f06b7c59e5856b
> Cr-Commit-Position: refs/heads/master@{#13958}
TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
Review-Url: https://codereview.webrtc.org/2298063003
Cr-Commit-Position: refs/heads/master@{#13988}
This is the stats collector for the new stats types, RTCStats[1] and
RTCStatsReport[2]. It so far only produces RTCPeerConnectionStats[3] as
an example of how it would collect stats. Each RTCStats subclass will
get a corresponding RTCStatsCollector::ProduceFooStats().
Stats reports are cached and returned as const references (ref
counting). This allows stats to be inspected by multiple observers and
across multiple threads. No copies will have to be made when surfacing
this to Blink or other places.
The current implementation of ProducePeerConnectionStats() only look at
existing DataChannels. This might be incorret if data channels can be
removed? Will investigate in a follow-up, crbug.com/636818.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#idl-def-rtcstats
[2] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
[3] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html#pcstats-dict*
BUG=chromium:627816, chromium:636818
Review-Url: https://codereview.webrtc.org/2242043002
Cr-Commit-Position: refs/heads/master@{#13979}
Reason for revert:
Breaks downstream.
Original issue's description:
> Remove the old AndroidVideoCapturer stack code.
>
> This code is no longer needed. Apps should be using the new API introduced here: https://codereview.webrtc.org/2127893002/
>
> Committed: https://crrev.com/1b365a8db070f9cdcbf35ec871f758dcd909e51d
> Cr-Commit-Position: refs/heads/master@{#13950}
TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@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/2291583002
Cr-Commit-Position: refs/heads/master@{#13958}
The old and new getStats are very different. This CL proposes rewriting
the new getStats from scratch with a bottom-up approach, starting with
the fundamental stats classes. This will allow cleaner and more
efficient code that is more aligned with the spec.
RTCStats and subclasses are the equivalent to RTCStats and RTCStats-
-derived dictionaries from the specs[1][2]. The dictionary members are
public member variables of type RTCStatsMember<T>, where T is one of the
supported types. All members derive from RTCStatsMemberInterface and
iteration of members is possible with RTCStats::Members().
The members are not stored in a map for performance and readability.
Type checking is supported with static class variables, kType.
Only the supported member types T are specialized and may be
instantiated, and sequences are supported with std::vector<...>. Type
checking is again supported with static class variables, kType.
RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id
to RTCStats-objects. RTCStatsReport is reference counted. It and its
contained stats may be destroyed on any thread. When the
RTCStatsCollector is added in a follow-up CL, it will return const
references to the RTCStatsReports. This means copies don't have to be
made for multiple stats observers or when jumping threads. In fact, no
copies of any stats will have to be made in surfacing stats to Blink.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary
[2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html
[3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
This adds the new folder webrtc/stats/, with target rtc_stats and binary
rtc_stats_unittests. Public api headers are placed in webrtc/api/ and
.cc files are placed in webrtc/stats/.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2241093002
Cr-Commit-Position: refs/heads/master@{#13879}
Reason for revert:
Breaks chromium.
Original issue's description:
> Add field_trial_default dependency to libjingle_peerconnection
>
> This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
>
> NOTRY=True
>
> Committed: https://crrev.com/a7a01df2aebe7108afad208ccd0341c2f0bc7b3b
> Cr-Commit-Position: refs/heads/master@{#13836}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org,kjellander@webrtc.org,arlolra@gmail.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2263063002
Cr-Commit-Position: refs/heads/master@{#13837}
This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
NOTRY=True
Review-Url: https://codereview.webrtc.org/2120673004
Cr-Commit-Position: refs/heads/master@{#13836}
New files, classes moved from statscollector_unittest.cc:
+webrtc/api/test/mock_peerconnection.h
for MockPeerConnectionFactory and MockPeerConnection
+webrtc/api/test/mock_webrtcsession.h
for MockWebRtcSession
+webrtc/media/base/test/mock_mediachannel.h
for MockVideoMediaChannel and MockVoiceMediaChannel
The webrtc/media/base/test folder is new.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2238933002
Cr-Commit-Position: refs/heads/master@{#13769}
stack will be removed soon in a separate CL. Constraints will not be supported
in the new implementation. Apps can request a format directly and the closest
supported format will be selected.
Changes needed from the apps:
1. Use the new createVideoSource without constraints.
2. Call startCapture manually.
3. Don't call videoSource.stop/restart, use startCapture/stopCapture instead.
R=magjed@webrtc.orgTBR=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/2127893002 .
Cr-Commit-Position: refs/heads/master@{#13504}
This interface and its implementations have been replaced by
rtc::RTCCertificateGeneratorInterface.
Removes dtlsidentitystore.h, updates .gyp/gn and removes old #includes.
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2034013003
Cr-Commit-Position: refs/heads/master@{#13432}
Reason for revert:
Issues fixed
Original issue's description:
> Revert of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2111823002/ )
>
> Reason for revert:
> Breaks downstream dependencies
>
> Original issue's description:
> > Combine webrtc/api/java/android and webrtc/api/java/src.
> >
> > It used to be that there was a Java api for devices not running Android
> > but that is no longer the case. I combined the directories and made
> > the folder structure chromium style.
> >
> > BUG=webrtc:6067
> > R=magjed@webrtc.org, tommi@webrtc.org
> >
> > Committed: ceefe20dd6
>
> TBR=magjed@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6067
>
> Committed: 9b0dc622d4TBR=magjed@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6067
Review-Url: https://codereview.webrtc.org/2111923003
Cr-Commit-Position: refs/heads/master@{#13363}
Reason for revert:
Breaks downstream dependencies
Original issue's description:
> Combine webrtc/api/java/android and webrtc/api/java/src.
>
> It used to be that there was a Java api for devices not running Android
> but that is no longer the case. I combined the directories and made
> the folder structure chromium style.
>
> BUG=webrtc:6067
> R=magjed@webrtc.org, tommi@webrtc.org
>
> Committed: ceefe20dd6TBR=magjed@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6067
Review URL: https://codereview.webrtc.org/2106333005 .
Cr-Commit-Position: refs/heads/master@{#13357}
It used to be that there was a Java api for devices not running Android
but that is no longer the case. I combined the directories and made
the folder structure chromium style.
BUG=webrtc:6067
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2111823002 .
Cr-Commit-Position: refs/heads/master@{#13356}
Relanding again after fixing issue with RTC_DCHECKs.
This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13305}
Reason for revert:
Broke video sending for iOS AppRTCDemo. To repro, run iOS AppRTCDemo in Release in loopback mode. The revision prior to this change worked.
Original issue's description:
> Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: 2d5491783aTBR=pthatcher@webrtc.org,deadbeef@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/2092273003
Cr-Commit-Position: refs/heads/master@{#13289}
