e680c83a41
This reverts commit 8848229234aae01ec19582ece7b748d557119d66. Reason for revert: break chromium compilation on iOS https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8915214519549611184/+/steps/compile/0/stdout Original change's description: > Add Video Bwe stats collection to DefaultVideoQualityAnalyzer. > > This CL adds the possibility to collect the following Video BWE stats: > - available_send_bandwidth > - transmission_bitrate > - retransmission_bitrate > - actual_encode_bitrate > - target_encode_bitrate > > Example of the output: > > RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond > RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond > RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond > RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond > RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond > RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond > RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond > RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond > RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond > RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond > > Bug: webrtc:10138 > Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Tommi <tommi@webrtc.org> > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@google.com> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27760} TBR=mbonadei@webrtc.org,mbonadei@google.com,ilnik@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org Change-Id: Ib0ef94331410d9b22b6425e4da412b75360fa2d9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134210 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27771}
62 lines
1.5 KiB
Python
62 lines
1.5 KiB
Python
include_rules = [
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"+third_party/libjpeg",
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"+third_party/libjpeg_turbo",
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"+call",
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"+common_audio",
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"+common_video",
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"+logging/rtc_event_log",
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"+media/base",
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"+media/engine",
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"+modules/audio_coding",
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"+modules/congestion_controller",
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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/congestion_controller/bbr",
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"+modules/rtp_rtcp",
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"+modules/utility",
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"+modules/video_capture",
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"+modules/video_coding",
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"+sdk",
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"+system_wrappers",
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"+third_party/libyuv",
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]
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specific_include_rules = {
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"gmock\.h": [
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"+testing/gmock/include/gmock",
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],
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"gtest\.h": [
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"+testing/gtest/include/gtest",
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],
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".*congestion_controller_feedback_fuzzer\.cc": [
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"+modules/congestion_controller/include/receive_side_congestion_controller.h",
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"+modules/pacing/packet_router.h",
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"+modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h",
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],
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".*mdns_parser_fuzzer\.cc": [
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"+p2p/base/mdns_message.h",
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],
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".*pseudotcp_parser_fuzzer\.cc": [
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"+p2p/base/pseudo_tcp.h",
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],
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".*stun_parser_fuzzer\.cc": [
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"+p2p/base/stun.h",
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],
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".*stun_validator_fuzzer\.cc": [
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"+p2p/base/stun.h",
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],
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".*test_peer\.(h|cc)": [
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"+pc",
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"+p2p",
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],
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".*network_emulation_pc_unittest\.cc": [
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"+pc/peer_connection_wrapper.h",
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"+pc/test/mock_peer_connection_observers.h",
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"+p2p/client/basic_port_allocator.h",
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],
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".*peer_connection_quality_test\.(h|cc)": [
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"+pc",
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]
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}
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