webrtc/api
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.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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
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.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
rtcerror_unittest.cc 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.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
rtpparameters_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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