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webrtc_m130/call
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Danil Chapovalov 50ca701e68 Mitigate scenario when VP9 HW encoder uses less spatial layers than configured 
log an error and reset chain instead of propagating oversized chain diff, thus avoid DCHECK in RtpDependencyDescriptorWriter.

Bug: webrtc:11999, chromium:1467865
Change-Id: If0d186ce7d3a108799ef2b42dd4a3791db33f1a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314501
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40503}
2023-08-02 11:54:46 +00:00
..
adaptation
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
test
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Propagate time of the last received packet with Timestamp type
2023-06-02 14:29:19 +00:00
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
audio_send_stream.h
Replace "rcvd" with "received" for readability
2023-04-24 15:30:07 +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
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
2022-03-09 13:23:21 +00:00
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
bitrate_estimator_tests.cc
Reland: FrameGeneratorCapturer: don't generate video before Start is called
2023-06-29 14:47:05 +00:00
BUILD.gn
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
call_config.cc
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
call_config.h
Introduce support for video packet batching.
2023-05-08 16:24:03 +00:00
call_factory.cc
Remove CoDel from webrtc::SimulatedNetwork.
2022-09-08 06:51:05 +00:00
call_factory.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
call_perf_tests.cc
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +00:00
call_unittest.cc
Use Timestamp type in RtpState struct
2023-04-27 11:24:38 +00:00
call.cc
Pass and process capture time through SendPacketObserver with Timestamp type
2023-06-22 17:16:41 +00:00
call.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
degraded_call.cc
Migrate PostTask+Wait to BlockingCall to avoid deadlock in DegradedCall.
2023-04-19 09:40:33 +00:00
degraded_call.h
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
fake_network_pipe_unittest.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
fake_network_pipe.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
fake_network_pipe.h
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
flexfec_receive_stream_impl.cc
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.h
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +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
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
OWNERS
Update OWNERS for call/
2022-06-03 12:01:46 +00:00
packet_receiver.h
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
rampup_tests.cc
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +00:00
rampup_tests.h
Stop overriding extensions in rampup tests
2023-01-25 13:18:49 +00:00
receive_stream.h
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +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
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
rtp_config.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
rtp_config.h
Introduce support for video packet batching.
2023-05-08 16:24:03 +00:00
rtp_demuxer_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
rtp_demuxer.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Support conversion of VP9 non-flexible mode to generic descriptor for non-layered streams only.
2023-06-20 10:52:59 +00:00
rtp_payload_params.cc
Mitigate scenario when VP9 HW encoder uses less spatial layers than configured
2023-08-02 11:54:46 +00:00
rtp_payload_params.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_stream_receiver_controller_interface.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller.cc
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller.h
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_transport_config.h
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
rtp_transport_controller_send_factory_interface.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
rtp_transport_controller_send_factory.h
Refactor some config plumbing in call/.
2022-11-16 09:18:40 +00:00
rtp_transport_controller_send_interface.h
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
rtp_transport_controller_send.cc
Reduce precision of RTT in RtrpTransportControllerSend
2023-05-19 10:38:37 +00:00
rtp_transport_controller_send.h
Cleanup RtcpReceiver from passing TransportFeedback via older interface
2023-05-22 10:14:04 +00:00
rtp_video_sender_interface.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +00:00
rtp_video_sender_unittest.cc
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
2023-06-01 07:51:56 +00:00
rtp_video_sender.cc
Recreate the stream when switching from standard to legacy API.
2023-07-26 13:48:41 +00:00
rtp_video_sender.h
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
2023-06-01 07:51:56 +00:00
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.cc
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
rtx_receive_stream.h
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
simulated_network_unittest.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.h
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +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
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2023-08-01T04:05:22).
2023-08-01 05:31:25 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay
2023-05-26 13:34:09 +00:00
video_receive_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +00:00
video_send_stream.cc
Cleanup usasge of ReportBlockData::report_block accessor
2023-05-05 09:56:30 +00:00
video_send_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +00:00