Reason for revert:
I unfortunately have to revert this since trybots are still trying to run these tests and now they fail because the tests aren't there :-/ Before relanding, can you work with engprod to make sure we don't run them?
Original issue's description:
> Removes a test suite that is no longer used or maintained.
>
> BUG=NONE
>
> Review-Url: https://codereview.webrtc.org/2716843002
> Cr-Commit-Position: refs/heads/master@{#16825}
> Committed: bb5136f940TBR=solenberg@webrtc.org,nisse@webrtc.org,henrika@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2721433002
Cr-Commit-Position: refs/heads/master@{#16845}
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}
It's currently not possible on Mac and iOS due to libtool. See webrtc:6418
for more info.
BUG=webrtc:6418
NOTRY=True
Review-Url: https://codereview.webrtc.org/2367313002
Cr-Commit-Position: refs/heads/master@{#16411}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
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}
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.
Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: eaa826c2eeTBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16196}
Use rtc::TimeMicros, and don't refer to ntp time.
BUG=webrtc:6977
Review-Url: https://codereview.webrtc.org/2633673002
Cr-Commit-Position: refs/heads/master@{#16138}
A decision has been made to not use prefix header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2590823002
Cr-Commit-Position: refs/heads/master@{#16056}
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2608353003
Cr-Commit-Position: refs/heads/master@{#16041}
Last references to the old code were fixed in Chromium with
https://codereview.chromium.org/2616873002/ and
https://codereview.chromium.org/2617363002/
Original issue's description:
> Revert of Remove webrtc/libjingle/{xmllite,xmpp} (patchset #1 id:1 of https://codereview.webrtc.org/2617443003/ )
>
> Reason for revert:
> Breaks Chromium FYI bots.
>
> tommi, please let me land this (I forgot to run them).
>
> Original issue's description:
> > Remove webrtc/libjingle/{xmllite,xmpp} as it's dead code.
> >
> > These sources have now been imported into Chromium's
> > src/third_party/libjingle_xmpp.
> >
> > BUG=webrtc:5539
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2617443003
> > Cr-Commit-Position: refs/heads/master@{#15910}
> > Committed: 1670b1fe6b
>
> TBR=tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5539
>
> Review-Url: https://codereview.webrtc.org/2618633003
> Cr-Commit-Position: refs/heads/master@{#15911}
> Committed: 60ef117be4TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5539
Review-Url: https://codereview.webrtc.org/2618693002
Cr-Commit-Position: refs/heads/master@{#15954}
Reason for revert:
Breaks Chromium FYI bots.
tommi, please let me land this (I forgot to run them).
Original issue's description:
> Remove webrtc/libjingle/{xmllite,xmpp} as it's dead code.
>
> These sources have now been imported into Chromium's
> src/third_party/libjingle_xmpp.
>
> BUG=webrtc:5539
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2617443003
> Cr-Commit-Position: refs/heads/master@{#15910}
> Committed: 1670b1fe6bTBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5539
Review-Url: https://codereview.webrtc.org/2618633003
Cr-Commit-Position: refs/heads/master@{#15911}
These sources have now been imported into Chromium's
src/third_party/libjingle_xmpp.
BUG=webrtc:5539
NOTRY=True
Review-Url: https://codereview.webrtc.org/2617443003
Cr-Commit-Position: refs/heads/master@{#15910}
This will make it easier for some downstream projects to control whether
or not to set ENABLE_EXTERNAL_AUTH, via the GN variable.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2608673002
Cr-Commit-Position: refs/heads/master@{#15894}
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}
This effectively reverts commit c3e1cabc696240e4b5a128653264785292878205
(https://codereview.webrtc.org/2589703002/).
The reason the test was failing before was missing resource
dependencies in the GN file. This is now fixed.
Furthermore, the test did not trigger the complexity adaptation that
it was supposed to test, since the hysteresis window of the bitrate
was not taken into account. This is also fixed.
Finally, a percent label was added to a printout, to match the same
printout in the other test.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2580383002
Cr-Commit-Position: refs/heads/master@{#15679}
BUG=webrtc:6649
- Supports Bluetooth Headset profile.
- Detects new BT headset:
+ enabled at start, and
+ powered on during active call.
- Enables/disables BT SCO channel when BT device is selected.
- Removes proximity sensor usage to avoid conflicts (will be added again later).
- Adds new (unused) APIs to explicitly select audio device.
- Starts routing audio to BT headset when enabled, i.e, BT is default.
Review-Url: https://codereview.webrtc.org/2501983002
Cr-Commit-Position: refs/heads/master@{#15610}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15453}
Reason for revert:
Deletion of transport.h broke downstream builds.
Going to reland with transport.h containing enums/etc.
Original issue's description:
> Refactoring that removes P2PTransport and DtlsTransport classes.
>
> Their base class, Transport, still exists, but it now has a more specific
> role: a helper class that applies TransportDescriptions. And is renamed
> to JsepTransport as a result.
>
> TransportController is now the entity primarily responsible for managing
> TransportChannels. It also starts storing pointers to the DTLS and ICE
> chanels separately, which will make it easier to remove
> TransportChannel/TransportChannelImpl in a subsequent CL.
>
> BUG=None
>
> Committed: https://crrev.com/bd28681d02dee8c185aeb39207e8154f0ad14a37
> Cr-Commit-Position: refs/heads/master@{#15450}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2553043004
Cr-Commit-Position: refs/heads/master@{#15452}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15450}
The CL fixes adds tests that fully test the functions that manipulate the cricket::VideoFormat<->AVCaptureDeviceFormat
relation.
BUG=webrtc:6680
Review-Url: https://codereview.webrtc.org/2526813002
Cr-Commit-Position: refs/heads/master@{#15444}
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}
It was incorrectly using a older version of the macro, which
wasn't discovered since the code wasn't built in WebRTC until now.
I moved webrtc/base/sigslottester.h from rtc_unittests into
rtc_base_test_utils instead to make it more usable.
BUG=webrtc:6821
Review-Url: https://codereview.webrtc.org/2551813002
Cr-Commit-Position: refs/heads/master@{#15422}
