882b32d00f
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95. Reason for revert: Revised codec matching to fix issue. Changes also back out some changes that should not have been included (using PayloadTypePicker for codec list merging). Original change's description: > Revert "Use PayloadTypePicker for video PT assignment" > > This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3. > > Reason for revert: Broke internal tests. > > Original change's description: > > Use PayloadTypePicker for video PT assignment > > > > This includes changes that change the order of codecs. > > It is preparatory to doing late assignment of video PTs. > > > > Bug: webrtc:360058654 > > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400 > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#43489} > > Bug: webrtc:360058654 > Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43490} Bug: webrtc:360058654 Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43554}
78 lines
1.8 KiB
Python
78 lines
1.8 KiB
Python
include_rules = [
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"+audio",
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"+logging/rtc_event_log",
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"+modules/async_audio_processing",
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"+modules/audio_coding",
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"+modules/audio_device",
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"+modules/audio_mixer",
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"+modules/audio_processing",
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"+modules/bitrate_controller",
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"+modules/congestion_controller",
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"+modules/video_coding",
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"+modules/pacing",
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"+modules/rtp_rtcp",
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"+modules/utility",
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"+system_wrappers",
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"+video",
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]
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specific_include_rules = {
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"video_receive_stream\.h": [
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"+common_video/frame_counts.h",
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],
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"video_send_stream\.h": [
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"+common_video",
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],
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"rtp_transport_controller_send_interface\.h": [
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"+common_video/frame_counts.h",
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],
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"call_perf_tests\.cc": [
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"+media/engine",
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],
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"simulated_network\.h": [
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"+test/network/simulated_network.h",
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],
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"rtp_payload_params\.cc": [
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"+common_video/generic_frame_descriptor",
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],
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"rtp_payload_params\.h": [
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"+common_video/generic_frame_descriptor",
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],
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"rtp_payload_params_unittest\.cc": [
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"+common_video/generic_frame_descriptor",
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],
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"rtp_video_sender\.cc": [
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"+common_video/frame_counts.h",
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"+common_video/generic_frame_descriptor",
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],
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"rtp_video_sender.h": [
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"+common_video/frame_counts.h",
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],
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"rtp_video_sender_unittest.cc": [
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"+common_video/frame_counts.h",
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"+common_video/generic_frame_descriptor",
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],
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"payload_type\.h": [
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"+media/base/codec.h",
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],
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"payload_type_picker\.h": [
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"+media/base/codec.h",
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"+media/base/media_constants.h",
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],
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"payload_type_picker\.cc": [
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"+media/base/codec.h",
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"+media/base/codec_comparators.h",
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"+media/base/media_constants.h",
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],
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"payload_type_picker_unittest\.cc": [
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"+media/base/codec.h",
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"+media/base/media_constants.h",
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],
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"call\.cc": [
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"+media/base/codec.h",
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],
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"fake_payload_type_suggester": [
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"+media/base/codec.h",
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]
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}
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