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webrtc_m130/call
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Erik Språng ef75ebef55 Directly include VideoBitrateAllocation in call/ targets 
Bug: webrtc:9271
Change-Id: Ie3f8d306bd7dac840c238acd26e36c0431b0f8c3
Reviewed-on: https://webrtc-review.googlesource.com/76941
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23246}
2018-05-15 15:35:02 +00:00
..
test
New api struct BitrateSettings.
2018-05-07 15:01:28 +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
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_send_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
audio_send_stream.h
Wire up MID send value to the PeerConnection API
2018-03-26 18:14:30 +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
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
bitrate_allocator_unittest.cc
Uses config struct with bitrate allocator.
2018-04-19 09:15:33 +00:00
bitrate_allocator.cc
Replace runtime thread checks by compile-time checks.
2018-04-27 07:51:55 +00:00
bitrate_allocator.h
Replace runtime thread checks by compile-time checks.
2018-04-27 07:51:55 +00:00
bitrate_constraints.h
New api struct BitrateSettings.
2018-05-07 15:01:28 +00:00
bitrate_estimator_tests.cc
Reland "Move creating encoder to VideoStreamEncoder."
2018-04-19 08:48:58 +00:00
BUILD.gn
Directly include VideoBitrateAllocation in call/ targets
2018-05-15 15:35:02 +00:00
call_config.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
call_config.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
call_perf_tests.cc
Directly include VideoBitrateAllocation in call/ targets
2018-05-15 15:35:02 +00:00
call_unittest.cc
Removed fake rtp transport controller send.
2018-02-21 13:39:16 +00:00
call.cc
Moves network control interface to API.
2018-05-09 11:01:36 +00:00
call.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
callfactory.cc
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
callfactory.h
Rename Call::Config to CallConfig, keep old name as alias.
2018-02-14 15:14:39 +00:00
degraded_call.cc
Fix race in DegradedCall::DestroyVideoSendStream
2018-03-16 10:23:47 +00:00
degraded_call.h
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Ensures that arrival is past send time in SimulatedNetwork.
2018-05-15 07:18:00 +00:00
fake_network_pipe.h
Extracting the simulation part of FakeNetworkPipe
2018-05-07 13:59:28 +00:00
flexfec_receive_stream_impl.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
flexfec_receive_stream_impl.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
flexfec_receive_stream.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
OWNERS
Added nisse@webrtc.org as owner in call.
2018-02-20 09:39:51 +00:00
rampup_tests.cc
Reland "Reland "Move rtp-specific config out of EncoderSettings.""
2018-04-05 14:30:09 +00:00
rampup_tests.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
receive_time_calculator_unittest.cc
Added receive time calculator under field trial.
2018-03-21 15:40:39 +00:00
receive_time_calculator.cc
Fix of ReceiveTimeCalculator field trial parsing.
2018-03-24 12:36:17 +00:00
receive_time_calculator.h
Added receive time calculator under field trial.
2018-03-21 15:40:39 +00:00
rtcp_demuxer_unittest.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtcp_demuxer.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtcp_demuxer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
New api struct BitrateSettings.
2018-05-07 15:01:28 +00:00
rtp_bitrate_configurator.cc
New api struct BitrateSettings.
2018-05-07 15:01:28 +00:00
rtp_bitrate_configurator.h
New api struct BitrateSettings.
2018-05-07 15:01:28 +00:00
rtp_config.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
rtp_config.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_demuxer_unittest.cc
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
2017-11-22 11:21:47 +00:00
rtp_demuxer.cc
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_demuxer.h
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +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_rtcp_demuxer_helper_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper.cc
Optional: Use nullopt and implicit construction in /call
2018-01-11 12:00:00 +00:00
rtp_rtcp_demuxer_helper.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_stream_receiver_controller.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_transport_controller_send_interface.h
New api struct BitrateSettings.
2018-05-07 15:01:28 +00:00
rtp_transport_controller_send.cc
Using shared task queue for congestion controller.
2018-05-07 15:54:38 +00:00
rtp_transport_controller_send.h
Using shared task queue for congestion controller.
2018-05-07 15:54:38 +00:00
rtx_receive_stream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtx_receive_stream.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ssrc_binding_observer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
video_config.cc
Add member VideoEncoderConfig::SdpVideoFormat.
2018-04-17 11:38:23 +00:00
video_config.h
Add member VideoEncoderConfig::SdpVideoFormat.
2018-04-17 11:38:23 +00:00
video_receive_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
video_receive_stream.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
video_send_stream.cc
Reland "Move creating encoder to VideoStreamEncoder."
2018-04-19 08:48:58 +00:00
video_send_stream.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00