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webrtc_m130/call
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Ilya Nikolaevskiy 2181228624 Revert "[VP9] Shift spatial layers on RTP level to always start from 0." 
This reverts commit 2e73a3d1e9298da6a010cd638f08f36abeba11e2.

Reason for revert: Fuzzer found some issues.

Original change's description:
> [VP9] Shift spatial layers on RTP level to always start from 0.
> 
> This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
> about enabled layers from encoder to packetizer.
> 
> Bug: webrtc:11319
> Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30428}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org

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

Bug: webrtc:11319
Change-Id: I27a7e82737fa604b8ab769ce6503fa93e46f4e86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168123
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30448}
2020-02-03 14:15:44 +00:00
..
adaptation
Move initial frame drop to overuse module
2020-02-03 11:56:31 +00:00
test
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +00:00
audio_receive_stream.cc
…
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
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +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
…
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
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
2019-12-07 00:54:26 +00:00
BUILD.gn
Reformat GN files.
2020-01-21 12:13:11 +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
Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
2020-01-07 13:02:52 +00:00
call_unittest.cc
Trials should always be populated in call config.
2019-12-03 10:34:55 +00:00
call.cc
Reduce some logging at INFO level by moving log statements
2019-12-12 21:54:06 +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
…
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
Concatenate string literals at compile time.
2020-01-14 14:47:48 +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
…
rtcp_packet_sink_interface.h
…
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
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +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
…
rtp_payload_params_unittest.cc
Pass picture_id from generic packetizer through codec-specific field
2020-01-28 19:26:28 +00:00
rtp_payload_params.cc
Revert "[VP9] Shift spatial layers on RTP level to always start from 0."
2020-02-03 14:15:44 +00:00
rtp_payload_params.h
Revert "Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused"
2020-01-07 19:16:48 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_rtcp_demuxer_helper.cc
…
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
…
rtp_stream_receiver_controller.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +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
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +00:00
rtp_transport_controller_send.cc
Reland "Adds trial to use correct overhead calculation in pacer."
2020-01-29 18:45:16 +00:00
rtp_transport_controller_send.h
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +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
Adds scenario test for transport wide feedback based retransmission.
2020-01-14 16:00:14 +00:00
rtp_video_sender.cc
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +00:00
rtp_video_sender.h
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +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
…
simulated_network_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
simulated_network.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
simulated_network.h
Exposing more features in the network emulation manager API.
2019-12-06 08:47:19 +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
…
syncable.cc
…
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