First approach to remove parts of the heavy load done for encoding, and
preparation for sending, from native audio thread to separate task queue.
With this change we will give the native input audio thread more time to
"relax" between successive audio captures.
Separate profiling done on Android has verified that the change works well;
the load is now redistributed and the load of the native AudioRecordThread
is reduced. Similar conclusions should be valid for all other OS:es as well.
BUG=NONE
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng
Review-Url: https://codereview.webrtc.org/2665693002
Cr-Commit-Position: refs/heads/master@{#17488}
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.
BUG=webrtc:6847, webrtc:7135
Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
Only one test case in it, testing an API which is deprecated.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2757213002
Cr-Commit-Position: refs/heads/master@{#17372}
These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name.
BUG=none
Review-Url: https://codereview.webrtc.org/2731993002
Cr-Commit-Position: refs/heads/master@{#17087}
The class basically implements a timer and can be replaced with a PostDelayedTask call down the line.
BUG=none
Review-Url: https://codereview.webrtc.org/2722613002
Cr-Commit-Position: refs/heads/master@{#16891}
The removed tests are covered by cases in call_perf_tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder.
BUG=webrtc:6904
Review-Url: https://codereview.webrtc.org/2579613003
Cr-Commit-Position: refs/heads/master@{#16217}
Because Voice Engine was the only user.
(We have tried to land this many times before. I'm hoping that this
time all external dependencies on these files will really be gone.)
BUG=none
Review-Url: https://codereview.webrtc.org/2622493002
Cr-Commit-Position: refs/heads/master@{#15978}
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}
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}
Methods to facilitate this are added to ChannelProxy and voe::Channel.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2378143004
Cr-Commit-Position: refs/heads/master@{#14707}
I added a few missing dependencies to the GN target of voice_engine while doing other
unrelated work. Currently GN's header include checker has the
following to say:
$ gn check out/gn_debug webrtc/voice_engine
ERROR at //webrtc/voice_engine/include/voe_network.h:38:11: Include not allowed.
#include "webrtc/transport.h"
^-----------------
It is not in any dependency of
//webrtc/voice_engine:voice_engine
The include file is in the target(s):
//webrtc:webrtc
which should somehow be reachable.
transport.h should probably move in to webrtc/api, since it is already
a pure virtual interface and is used in quite a few places.
BUG=webrtc:5589
NOTRY=True
Review-Url: https://codereview.webrtc.org/2421483002
Cr-Commit-Position: refs/heads/master@{#14633}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
This is done to ensure GN targets are placed in the same directory as of the source files.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
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}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Original-Commit-Position: refs/heads/master@{#14191}
Cr-Commit-Position: refs/heads/master@{#14198}
Reason for revert:
Breaks downstream code
Original issue's description:
> Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
>
> Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
>
> Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
>
> BUG=
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
> Cr-Commit-Position: refs/heads/master@{#14191}
TBR=kwiberg@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/2336123002
Cr-Commit-Position: refs/heads/master@{#14193}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Commit-Position: refs/heads/master@{#14191}
audio_coding should be in public_deps, in order to a define used by
voe_output_test.ccto be exported, as is done in GYP
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2321783003
Cr-Commit-Position: refs/heads/master@{#14161}
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}
Remove //build/config/sanitizers:deps as a dependency for
all rtc_executable targets and add it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2308553002
Cr-Commit-Position: refs/heads/master@{#14048}
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}
