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webrtc_m130/api
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Niels Möller d5696fb8f5 Add video support to LoopbackMediaTransport 
Bug: webrtc:9719
Change-Id: I568da8720377342cf44ee8caa316e14b4cd8beba
Reviewed-on: https://webrtc-review.googlesource.com/c/111960
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25826}
2018-11-28 15:34:20 +00:00
..
audio
AEC3: Fix ENR in the dominant nearend detection
2018-11-28 09:23:34 +00:00
audio_codecs
Adds OnReceivedUplinkAllocation method to AudioEncoder.
2018-11-21 20:46:01 +00:00
call
Moves BitrateAllocationUpdate to api.
2018-11-21 19:59:55 +00:00
crypto
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
ortc
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
stats
Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
2018-11-27 15:10:09 +00:00
test
Add video support to LoopbackMediaTransport
2018-11-28 15:34:20 +00:00
transport
Increasing visibility of api/transport build targets.
2018-11-15 09:05:53 +00:00
units
Adds shared base class for data units.
2018-11-19 12:41:33 +00:00
video
Delete method EncodedFrame::GetBitstream, part 1
2018-11-28 14:52:32 +00:00
video_codecs
Remove SetChannelParameters function from API classes.
2018-11-22 11:12:10 +00:00
array_view_unittest.cc
ArrayView, adding ctor for fixed-size views of const(expr) std::array.
2018-05-15 13:49:02 +00:00
array_view.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
asyncresolverfactory.h
Support domain name ICE candidates
2018-08-24 04:54:43 +00:00
audio_options.cc
Add PeerConnection option to configure minimum audio jitter buffer delay.
2018-11-27 19:49:48 +00:00
audio_options.h
Add PeerConnection option to configure minimum audio jitter buffer delay.
2018-11-27 19:49:48 +00:00
bitrate_constraints.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
BUILD.gn
Decouple //rtc_base:rtc_base_tests_utils from gunit.
2018-11-23 12:52:46 +00:00
candidate.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
candidate.h
Export symbols needed by the Chromium component build (part 2).
2018-10-15 19:52:31 +00:00
create_peerconnection_factory.cc
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
create_peerconnection_factory.h
Fix wrong forward declaration namespace.
2018-11-27 08:20:05 +00:00
cryptoparams.h
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
2018-10-11 23:09:07 +00:00
datachannelinterface.cc
Enabling clang::find_bad_constructs for libjingle_peerconnection_api.
2018-07-19 09:17:10 +00:00
datachannelinterface.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
DEPS
Move webrtc::CreatePeerConnectionFactory definition next to decl.
2018-11-22 09:07:51 +00:00
dtmfsenderinterface.h
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
fec_controller.h
Revert "Revert "Enables PeerConnectionFactory using external fec controller""
2018-02-20 12:41:55 +00:00
jsep.cc
Move SdpType from/to string definition close to declaration.
2018-10-12 09:59:40 +00:00
jsep.h
Export symbols needed by the Chromium component build (part 6).
2018-10-23 06:48:51 +00:00
jsepicecandidate.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
jsepicecandidate.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
jsepsessiondescription.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
media_transport_interface.cc
Add video support to LoopbackMediaTransport
2018-11-28 15:34:20 +00:00
media_transport_interface.h
Add video support to LoopbackMediaTransport
2018-11-28 15:34:20 +00:00
mediaconstraintsinterface.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
mediaconstraintsinterface.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
mediastreaminterface.cc
AudioSource allows implementations to return settings
2018-11-13 16:30:09 +00:00
mediastreaminterface.h
AudioSource allows implementations to return settings
2018-11-13 16:30:09 +00:00
mediastreamproxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamtrackproxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediatypes.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediatypes.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
notifier.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Add owners for media_transport_interface
2018-11-08 17:45:39 +00:00
peerconnectionfactoryproxy.h
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnectioninterface.cc
Compute RTCConnectionState and RTCIceConnectionState.
2018-10-22 11:33:17 +00:00
peerconnectioninterface.h
Add PeerConnection option to configure minimum audio jitter buffer delay.
2018-11-27 19:49:48 +00:00
peerconnectionproxy.h
Revert "Replace the IceConnectionState implementation."
2018-11-23 16:19:05 +00:00
proxy.cc
Make member internal::SynchronousMethodCall::e_ a non-pointer.
2018-11-15 10:42:36 +00:00
proxy.h
Make member internal::SynchronousMethodCall::e_ a non-pointer.
2018-11-15 10:42:36 +00:00
refcountedbase.h
New classes RefCounter and RefCountedBase.
2017-10-23 11:46:47 +00:00
rtcerror_unittest.cc
Remove all remaining non-test uses of std::stringstream.
2018-09-13 08:52:05 +00:00
rtcerror.cc
Remove all remaining non-test uses of std::stringstream.
2018-09-13 08:52:05 +00:00
rtcerror.h
Remove all remaining non-test uses of std::stringstream.
2018-09-13 08:52:05 +00:00
rtceventlogoutput.h
Move RtcEventLogOutput to api/
2017-10-06 13:58:14 +00:00
rtp_headers.cc
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
rtp_headers.h
Change HdrMetadataExtension to ColorSpaceExtension
2018-11-27 14:05:31 +00:00
rtpparameters_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpparameters.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
rtpparameters.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
rtpreceiverinterface.cc
Injects FrameEncryptorInterface into RtpSender.
2018-08-30 00:33:54 +00:00
rtpreceiverinterface.h
Injects FrameEncryptorInterface into RtpSender.
2018-08-30 00:33:54 +00:00
rtpsenderinterface.cc
Add support for send_encodings parameters in addTransceiver
2018-10-01 22:56:30 +00:00
rtpsenderinterface.h
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
rtptransceiverinterface.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
rtptransceiverinterface.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
scoped_refptr.h
Move rtc::scoped_refptr to api/.
2018-11-19 16:13:16 +00:00
setremotedescriptionobserverinterface.h
Reland "SetRemoteDescriptionObserverInterface added."
2017-11-23 19:59:48 +00:00
statstypes.cc
Reimplement rtc::ToString and rtc::FromString without streams.
2018-08-16 16:14:01 +00:00
statstypes.h
Revert "Add framesRendered to StatsReport"
2018-07-27 14:53:07 +00:00
turncustomizer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
umametrics.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
videosourceproxy.h
Replace rtc::Optional with absl::optional in api
2018-06-21 12:50:03 +00:00