admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/call
History
Erik Språng 17f82cfc68 Verifies trials are populated when creating a Call. 
This check just makes it more clear what the expectations are.

Pululating trials was made mandatory in an earlier CL, but if you don't
populate this field it will trigger a DCHECK at lower layer where we're
actually trying to parse an experiment. That is confusing and
misleading.

Bug: None
Change-Id: I1f520841a5a3b911048c8ee6d309eb7bb179e037
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161301
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30005}
2019-12-04 13:36:02 +00:00
..
test
Cleanup of feedback observer interface
2019-10-30 07:50:29 +00:00
audio_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_receive_stream.h
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_send_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_send_stream.h
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_state.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_state.h
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
bitrate_allocator_unittest.cc
Propagating TargetRate struct to BitrateAllocator.
2019-09-19 14:03:04 +00:00
bitrate_allocator.cc
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_allocator.h
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_estimator_tests.cc
Hide deprecated SingleThreadedTaskQueueForTest behind an accessor
2019-10-22 11:57:49 +00:00
BUILD.gn
VideoReceiveStream: Enable encoded frame sink.
2019-12-03 15:55:04 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Adds injectable trials from peerconnection down to transport controller.
2019-11-21 12:41:45 +00:00
call_factory.cc
DegradedCall: fake network using TaskQueue instead of ProcessThread
2019-08-06 15:05:30 +00:00
call_factory.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
call_perf_tests.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
call_unittest.cc
Trials should always be populated in call config.
2019-12-03 10:34:55 +00:00
call.cc
Verifies trials are populated when creating a Call.
2019-12-04 13:36:02 +00:00
call.h
Remove MediaTransport from Call.
2019-08-08 10:58:57 +00:00
degraded_call.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
degraded_call.h
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
fake_network_pipe.cc
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
fake_network_pipe.h
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
flexfec_receive_stream_impl.cc
Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc
2019-08-21 09:45:21 +00:00
flexfec_receive_stream_impl.h
Injecting Clock in video receive.
2019-03-04 21:53:57 +00:00
flexfec_receive_stream_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
flexfec_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h
Format almost everything.
2019-07-08 13:45:15 +00:00
OWNERS
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
packet_receiver.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
rampup_tests.cc
Do not stop SingleThreadedTaskQueueForTestingTest near the end of the tests
2019-10-31 13:07:30 +00:00
rampup_tests.h
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
receive_time_calculator_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
receive_time_calculator.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
receive_time_calculator.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_bitrate_configurator_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_bitrate_configurator.cc
Remove webrtc::MinPositive from api/.
2019-09-18 12:52:09 +00:00
rtp_bitrate_configurator.h
Remove api/bitrate_constraints.h.
2019-09-18 06:37:58 +00:00
rtp_config.cc
Negotiate use of RTCP loss notification feedback (LNTF)
2019-05-24 12:44:14 +00:00
rtp_config.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_demuxer_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
rtp_demuxer.cc
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +00:00
rtp_demuxer.h
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +00:00
rtp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_payload_params.cc
Add VideoCodecType::kVideoCodecAV1 value
2019-11-27 10:18:45 +00:00
rtp_payload_params.h
Reland "Populate the GFD-00 for H264 and generic codecs."
2019-06-14 14:47:06 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_rtcp_demuxer_helper.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
rtp_stream_receiver_controller_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
rtp_stream_receiver_controller.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
rtp_transport_controller_send_interface.h
Cleanup of TransportFeedbackAdapter.
2019-11-01 11:55:16 +00:00
rtp_transport_controller_send.cc
Trials should always be populated in call config.
2019-12-03 10:34:55 +00:00
rtp_transport_controller_send.h
Trials should always be populated in call config.
2019-12-03 10:34:55 +00:00
rtp_video_sender_interface.h
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +00:00
rtp_video_sender_unittest.cc
TimeController: Rename Sleep to AdvanceTime.
2019-12-03 16:08:54 +00:00
rtp_video_sender.cc
Add string<->VideoCodecType conversion for all codec types.
2019-11-27 14:15:07 +00:00
rtp_video_sender.h
Cleanup of feedback observer interface
2019-10-30 07:50:29 +00:00
rtx_receive_stream_unittest.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
simulated_network_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
simulated_network.cc
Adds CoDel implementation to network simulation.
2019-02-25 09:54:03 +00:00
simulated_network.h
Format almost everything.
2019-07-08 13:45:15 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
ssrc_binding_observer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.h
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
video_receive_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
video_receive_stream.h
VideoReceiveStream: Enable encoded frame sink.
2019-12-03 15:55:04 +00:00
video_send_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
video_send_stream.h
Delete media transport integration.
2019-11-26 19:19:36 +00:00