zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
f7cf133ad5f0a1341a6e31d58ffea8e63e8b22e4
platform-external-webrtc/call
History
Guido Urdaneta 61b2993aaf Revert "[Adaptation] Remove QualityScalerResource when disabled." 
This reverts commit ba8abbb630cdd9d05e22c830d0845e920762850d.

Reason for revert: Suspect of causing Chormium trybots to fail, preventing rolls. Will reland if the revert does not fix it.

Original change's description:
> [Adaptation] Remove QualityScalerResource when disabled.
> 
> Bug: webrtc:11843
> Change-Id: I2d3e40356c266f189db0242f3c7590e6d83e4456
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181369
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31924}

TBR=ilnik@webrtc.org,eshr@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11843
Change-Id: Idc3950be209c6edce0dbe72d98c9b4becae0049f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181880
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31952}
2020-08-17 13:46:30 +00:00
..
adaptation
Revert "[Adaptation] Remove QualityScalerResource when disabled."
2020-08-17 13:46:30 +00:00
test
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +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
Insert audio frame transformer between depacketizer and decoder.
2020-04-01 08:15:53 +00:00
audio_send_stream.cc
Log audio network adaptor and DSCP in AudioSendStream.
2020-08-13 14:05:46 +00:00
audio_send_stream.h
negotiate RED codec for audio
2020-06-25 06:24:18 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +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
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
bitrate_allocator_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
bitrate_allocator.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +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
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
BUILD.gn
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Remove deprecated constant.
2020-04-27 10:32:45 +00:00
call_factory.cc
Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread
2020-06-30 15:38:35 +00:00
call_factory.h
Add SharedModuleThread class to share a module thread across Call instances.
2020-05-25 17:21:56 +00:00
call_perf_tests.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
call_unittest.cc
[Adaptation] Multi-processor support for injected Resources.
2020-07-02 10:28:11 +00:00
call.cc
[Adaptation] Multi-processor support for injected Resources.
2020-07-02 10:28:11 +00:00
call.h
Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread
2020-06-30 15:38:35 +00:00
degraded_call.cc
[Adaptation] Adding adaptation resources from Call.
2020-06-11 12:43:21 +00:00
degraded_call.h
[Adaptation] Adding adaptation resources from Call.
2020-06-11 12:43:21 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
fake_network_pipe.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
fake_network_pipe.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
flexfec_receive_stream_impl.cc
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
flexfec_receive_stream_impl.h
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +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
Add terelius as OWNER in call/
2020-03-23 09:55:34 +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
Set up a new rtc::Thread instance per test.
2020-05-15 09:13:02 +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 newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
receive_time_calculator.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_config.cc
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_config.h
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_demuxer_unittest.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.h
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Remove framemarking RTP extension.
2020-06-15 11:18:00 +00:00
rtp_payload_params.cc
Propagate active decode targets bitmask into DependencyDescriptor
2020-06-29 12:54:43 +00:00
rtp_payload_params.h
Delete field trial WebRTC-GenericDescriptor
2020-06-03 13:00:30 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
rtp_stream_receiver_controller.h
Rename CriticalSection to RecursiveCriticalSection.
2020-07-17 09:19:50 +00:00
rtp_transport_controller_send_interface.h
Insert frame transformer between Encoded and Packetizer.
2020-02-28 07:43:13 +00:00
rtp_transport_controller_send.cc
Optionally allows TaskQueuePacedSender to coalesce send events.
2020-05-19 17:23:30 +00:00
rtp_transport_controller_send.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +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
Stop mentioning RTPFragmentationHeader in call/
2020-08-12 13:59:38 +00:00
rtp_video_sender.cc
Stop mentioning RTPFragmentationHeader in call/
2020-08-12 13:59:38 +00:00
rtp_video_sender.h
Stop mentioning RTPFragmentationHeader in call/
2020-08-12 13:59:38 +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
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
simulated_network.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
syncable.cc
…
syncable.h
Add periodic logging of sync delays.
2020-02-11 09:43:49 +00:00
video_receive_stream.cc
Add commas between codec parameters in VideoReceiveStream logging.
2020-03-09 02:45:34 +00:00
video_receive_stream.h
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
video_send_stream.cc
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
video_send_stream.h
[Adaptation] Adding adaptation resources from Call.
2020-06-11 12:43:21 +00:00