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webrtc/test
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Mirko Bonadei 6f440ed5b5 Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." 
This reverts commit 07efe436c9.

Reason for revert: Breaks downstream project.

cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).


Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
> 
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
>   under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
> 
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}

TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com

Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:14 +00:00
..
android Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199) 2018-02-22 13:58:58 +00:00
fuzzers Add AGC1 fuzzer 2018-06-21 13:09:03 +00:00
gl Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
ios Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
linux Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mac Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
testsupport Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
win Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8." 2018-06-21 13:41:14 +00:00
call_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
call_test.h Reland "Move creating encoder to VideoStreamEncoder." 2018-04-19 08:48:58 +00:00
configurable_frame_size_encoder.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
configurable_frame_size_encoder.h Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
constants.cc Remove voe_auto_test and add new tests to cover the missing cases. 2017-09-15 16:56:08 +00:00
constants.h Remove voe_auto_test and add new tests to cover the missing cases. 2017-09-15 16:56:08 +00:00
DEPS Reland "Add multiplex case to webrtc_perf_tests" 2018-03-10 01:21:04 +00:00
direct_transport.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
direct_transport.h Moving demux from FakeNetworkPipe to DirectTransport. 2018-04-25 10:13:03 +00:00
direct_transport_unittest.cc Moving demux from FakeNetworkPipe to DirectTransport. 2018-04-25 10:13:03 +00:00
drifting_clock.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
drifting_clock.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
encoder_proxy_factory.h Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
encoder_settings.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
encoder_settings.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_decoder.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_decoder.h Delete pre_decode_callback. 2018-06-20 07:04:09 +00:00
fake_encoder.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_encoder.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_texture_frame.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fake_texture_frame.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fake_videorenderer.h New file api/video/BUILD.gn 2018-05-14 06:57:38 +00:00
field_trial.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
field_trial.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
frame_generator.cc Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
frame_generator.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
frame_generator_capturer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
frame_generator_capturer.h Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
frame_generator_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
frame_utils.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
frame_utils.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
function_video_decoder_factory.h Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} 2018-05-31 11:48:17 +00:00
function_video_encoder_factory.h Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} 2018-05-31 11:48:17 +00:00
gmock.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gtest.h Reland "Adding gtest-spi.h in webrtc/test/gtest.h" 2018-04-05 08:21:23 +00:00
layer_filtering_transport.cc Delete enum RtpVideoCodecTypes, replaced with VideoCodecType. 2018-06-04 11:53:17 +00:00
layer_filtering_transport.h Moving demux from FakeNetworkPipe to DirectTransport. 2018-04-25 10:13:03 +00:00
mock_audio_decoder.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_audio_decoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_audio_decoder_factory.h Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
mock_audio_encoder.cc Removed Die mock from MockAudioEncoder 2018-02-22 12:53:38 +00:00
mock_audio_encoder.h Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
mock_audio_encoder_factory.h Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
mock_transport.cc Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" 2018-03-09 16:04:35 +00:00
mock_transport.h Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" 2018-03-09 16:04:35 +00:00
null_platform_renderer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
null_transport.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
null_transport.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Remove pbos@webrtc.org from all OWNERS. 2017-11-01 08:03:46 +00:00
rtcp_packet_parser.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtcp_packet_parser.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_file_reader.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_file_reader.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_file_reader_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_file_writer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_file_writer.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_file_writer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_observer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
run_loop.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
run_loop.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
run_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
run_test.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
single_threaded_task_queue.cc Fixing typo in a comment. 2017-12-14 09:07:31 +00:00
single_threaded_task_queue.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
single_threaded_task_queue_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
statistics.cc Reland "Updated analysis in videoprocessor." 2018-01-18 08:37:27 +00:00
statistics.h Reland "Updated analysis in videoprocessor." 2018-01-18 08:37:27 +00:00
test_main.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
vcm_capturer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
vcm_capturer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
video_capturer.cc Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
video_capturer.h Replace rtc::Optional with absl::optional in test and rtc_tools 2018-06-18 13:15:23 +00:00
video_codec_settings.h Delete RTP-specific values from the VideoCodecType enum. 2018-06-07 07:49:27 +00:00
video_renderer.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
video_renderer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00