VideoFrameBuffer is currently hard coded to be either I420 or Native.
This CL makes VideoFrameBuffer more generic by moving the I420 specific
functions into their own class, and adds an enum tag that represents the
format and storage type of the buffer. Each buffer type is then
represented as a subclass. See webrtc/api/video/video_frame_buffer.h for
more info.
This CL also adds support for representing I444 in VideoFrameBuffer
using the new interface. Possible future buffer type candidates are
RGB and NV12.
BUG=webrtc:7632
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2847383002
Cr-Commit-Position: refs/heads/master@{#18098}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
This target keeps track of .h the files under webrtc/modules/include/
that are not part of any target.
If a .h file is not part of a target the 'gn check' utility is not
able to spot if a target is missing a dependency because even if
it parses '#include' directives it is not able to find a target that
contains these headers.
BUG=webrtc:7513
NOTRY=True
Review-Url: https://codereview.webrtc.org/2838873002
Cr-Commit-Position: refs/heads/master@{#17880}
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}
Introduce new small header-only targets in system_wrappers:
:cpu_features_api
:field_trial_api
:metrics_api
to untangle and optimize dependencies but still satisfy GN check.
In webrtc/p2p, previously uncovered header "base/fakecandidatepair.h"
is added to :p2p_test_utils target.
Refactor system_wrappers so 'rtc_p2p' can depend on only
system_wrappers:field_trial_api instead of all of system_wrappers
(which led to a breakage in Chromium that called for the revert of
https://codereview.webrtc.org/2735583002).
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2739863002
Cr-Commit-Position: refs/heads/master@{#17812}
Create the SrtpTransportInterface, a subclass of RtpTransportInterface, which
allows the user to set the send and receive keys. The functionalities are
implemented inside the RtpTransportAdapters on top of BaseChannel.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2714813004
Cr-Commit-Position: refs/heads/master@{#17023}
Reason for revert:
Fails Chromium builds:
b/c/b/linux/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //third_party/webrtc/api/BUILD.gn:186:5: Can't load input file.
"//webrtc/test:test_support",
^-------------------------
Original issue's description:
> GN: Include webrtc/api targets even if rtc_include_tests=false
>
> The main purpose with the rtc_include_tests GN variable is to avoid
> generating and compiling all the test targets.
> Some of our examples have dependencies on the test headers in API,
> so therefore this change is relaxing that condition.
>
> BUG=webrtc:6828
> NOTRY=True
> TBR=ehmaldonado@webrtc.org,
>
> Review-Url: https://codereview.webrtc.org/2725053008
> Cr-Commit-Position: refs/heads/master@{#16989}
> Committed: a769ceba65TBR=ehmaldonado@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/2728073002
Cr-Commit-Position: refs/heads/master@{#16990}
The main purpose with the rtc_include_tests GN variable is to avoid
generating and compiling all the test targets.
Some of our examples have dependencies on the test headers in API,
so therefore this change is relaxing that condition.
BUG=webrtc:6828
NOTRY=True
TBR=ehmaldonado@webrtc.org,
Review-Url: https://codereview.webrtc.org/2725053008
Cr-Commit-Position: refs/heads/master@{#16989}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
This avoids adding an additional test target. Plus, everything in
rtc_api_unittests is (and likely will be) simple utility classes akin to
what's already being tested in rtc_unittests.
BUG=None
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2709573003
Cr-Commit-Position: refs/heads/master@{#16819}
This utility class can be used to represent either an error or a
successful return value. Follows the pattern of StatusOr in the protobuf
library.
This will be used by ORTC factory methods; for instance, CreateRtpSender
will either return an RtpSender or an error if the parameters are
invalid or some other failure occurs.
This CL also moves RTCError classes to a separate file, and adds tests
that were missing before.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2692723002
Cr-Commit-Position: refs/heads/master@{#16659}
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}
