admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/rtc_tools/DEPS

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

48 lines
1.5 KiB
Plaintext
Raw Normal View History

Reland "Add check_deps rules in DEPS files." Relanding https://codereview.webrtc.org/1796413002/ without the change to the openmax_dl include path (which broke downstream code). TBR=tommi@webrtc.org BUG=webrtc:5623 TESTED=Passing runs using: buildtools/checkdeps/checkdeps.py --root=. talk buildtools/checkdeps/checkdeps.py --root=. webrtc Review URL: https://codereview.webrtc.org/1804333002 . Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 12:05:36 +01:00
include_rules = [
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"+call",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"+common_audio",
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"+common_video",
"+logging/rtc_event_log",
Add rtp_generator utility to rtc_tools. This CL introduces a new rtp_generator tool that can be utilized to generate .rtpdump files that can be replayed by the video_replayer. This allows automated generation of corpus material for the new WebRTC RTP fuzzers in addition to allowing anyone who is experimenting with a new RTP feature to quickly debug issues. It can be used as follows: ./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump ./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump It works by generating squares randomly on the screen for a given duration. This initial version is very limited and doesn't support FEC, RED and other configurations. I plan to extend it to support these in future CLs. Bug: webrtc:10117 Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51 Reviewed-on: https://webrtc-review.googlesource.com/c/119964 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26517}
2019-02-01 10:00:05 -08:00
"+media",
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"+modules/audio_device",
"+modules/audio_coding/audio_network_adaptor",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"+modules/audio_coding/neteq/include",
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"+modules/audio_coding/neteq/tools",
"+modules/audio_processing",
"+modules/bitrate_controller",
Removes analyzer dependency on legacy congestion controller. Bug: webrtc:9586 Change-Id: Ic1f2445d6432202aeba9164acd49b75261e91aa0 Reviewed-on: https://webrtc-review.googlesource.com/c/105107 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25183}
2018-10-15 18:46:51 +02:00
"+modules/remote_bitrate_estimator",
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"+modules/congestion_controller",
"+modules/pacing",
"+modules/rtp_rtcp",
"+system_wrappers",
"+p2p",
Including libyuv headers using fully qualified paths. Using fully qualified paths to include libyuv headers allows WebRTC to avoid to rely on the //third_party/libyuv:libyuv_config target to set the -I compiler flag. Today some WebRTC targets depend on //third_party/libyuv only to include //third_party/libyuv:libyuv_config but with fully qualified paths this should not be needed anymore. A follow-up CL will remove //third_party/libyuv from some targets that don't need it because they are not including libyuv headers. Bug: webrtc:8605 Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/28220 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21209}
2017-12-11 09:32:13 +01:00
"+third_party/libyuv",
Move EncoderStreamFactory into own file This cl/ is a NOP refactoring, moving the EncoderStreamFactory from within webrtc_video_engine.cc into own file in video/. simulcast.cc is collateral. Bug: webrtc:14451 Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38224}
2022-09-27 16:53:04 +02:00
"+video/config",
Reland "Add check_deps rules in DEPS files." Relanding https://codereview.webrtc.org/1796413002/ without the change to the openmax_dl include path (which broke downstream code). TBR=tommi@webrtc.org BUG=webrtc:5623 TESTED=Passing runs using: buildtools/checkdeps/checkdeps.py --root=. talk buildtools/checkdeps/checkdeps.py --root=. webrtc Review URL: https://codereview.webrtc.org/1804333002 . Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 12:05:36 +01:00
]
Add YUV to IVF video converter util Bug: webrtc:10138 Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 15:03:45 +01:00
specific_include_rules = {
".*ivf_converter\.cc": [
"+absl/debugging/failure_signal_handler.h",
"+absl/debugging/symbolize.h",
"+modules/video_coding/codecs/vp8/include/vp8.h",
"+modules/video_coding/codecs/vp9/include/vp9.h",
"+modules/video_coding/include/video_error_codes.h",
"+modules/video_coding/utility/ivf_file_writer.h",
"+modules/video_coding/codecs/h264/include/h264.h",
],
video_replay: add support for IVF file output adding a -decoder_ivf_filename command line flag. BUG=none Change-Id: I895a6bf8093d5f36c17462d97240b17ada4dc9f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201560 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33077}
2021-01-26 18:05:56 +01:00
".*video_replay\.cc": [
"+modules/video_coding/utility/ivf_file_writer.h",
video_replay tool fix/update Change-Id: I188ed4c7cebc5cafb31e01a38e0460cde65fefa1 Bug: none Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341800 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41926}
2024-03-04 11:48:58 +01:00
"+absl/strings/str_split.h",
video_replay: add support for IVF file output adding a -decoder_ivf_filename command line flag. BUG=none Change-Id: I895a6bf8093d5f36c17462d97240b17ada4dc9f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201560 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33077}
2021-01-26 18:05:56 +01:00
],
[rct_tools/video_encoder] Add video encoder tool This CL adds a new video encode tool that supports to encode video at specified codec, scalability mode, resolution, frame rate, bitrate, key frame interval and the number of encoding frames. The video encoder accepts video frames from `FrameGeneratorInterface`, and supports `SquareFrameGenerator`, `SlideFrameGenerator` and `IvfFileFrameGenerator`. All the encoded bitstreams are wrote into ivf output files. The purposes of this video encoder tool are: 1. Check the functionalities of video codecs and scalability modes. 2. Optimize video quality at different encode setting. 3. Fine tune the bitrate controller. 4. Compare the quality of different codecs at the same setting. 5. And more. TESTS: Run the tool at 1280x720, 30fps, 2000kbps, 100 GOP, 300 frames: vp8 [L1T1 L1T3] h264 [L1T1 L1T3] vp9 [L1T1 L1T3 L3T3_KEY] av1 [L1T1 L1T3 L3T3_KEY] Bug: webrtc:15210 Change-Id: I3b0e463cf3236cd9a481fbab5688643c203958da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307361 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40267}
2023-06-12 19:36:03 +08:00
".*video_encoder\.cc": [
"+modules/video_coding/codecs/av1/av1_svc_config.h",
"+modules/video_coding/include/video_codec_interface.h",
"+modules/video_coding/svc/scalability_mode_util.h",
[rtc_tools/video_encoder] Output ivf for all SVC decode targets This CL extracts the ivf file writer from `TestEncodedImageCallback` into separate .cc|.h files. Improve the `EncodedImageFileWriter` to support SVC that output ivf for all decode targets. EXAMPLE: Encode with VP9 L3T3_KEY, the outputs: output-VP9-L3T3_KEY-L0T0.ivf output-VP9-L3T3_KEY-L0T1.ivf output-VP9-L3T3_KEY-L0T2.ivf output-VP9-L3T3_KEY-L1T0.ivf output-VP9-L3T3_KEY-L1T1.ivf output-VP9-L3T3_KEY-L1T2.ivf output-VP9-L3T3_KEY-L2T0.ivf output-VP9-L3T3_KEY-L2T1.ivf output-VP9-L3T3_KEY-L2T2.ivf Bug: webrtc:15210 Change-Id: Iba46c897a7b783bb4b79ec18715e901476cb9f55 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309280 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com> Cr-Commit-Position: refs/heads/main@{#40363}
2023-06-19 09:36:42 +08:00
],
".*encoded_image_file_writer\.(cc|h)": [
"+modules/video_coding/include/video_codec_interface.h",
"+modules/video_coding/svc/scalability_mode_util.h",
[rct_tools/video_encoder] Add video encoder tool This CL adds a new video encode tool that supports to encode video at specified codec, scalability mode, resolution, frame rate, bitrate, key frame interval and the number of encoding frames. The video encoder accepts video frames from `FrameGeneratorInterface`, and supports `SquareFrameGenerator`, `SlideFrameGenerator` and `IvfFileFrameGenerator`. All the encoded bitstreams are wrote into ivf output files. The purposes of this video encoder tool are: 1. Check the functionalities of video codecs and scalability modes. 2. Optimize video quality at different encode setting. 3. Fine tune the bitrate controller. 4. Compare the quality of different codecs at the same setting. 5. And more. TESTS: Run the tool at 1280x720, 30fps, 2000kbps, 100 GOP, 300 frames: vp8 [L1T1 L1T3] h264 [L1T1 L1T3] vp9 [L1T1 L1T3 L3T3_KEY] av1 [L1T1 L1T3 L3T3_KEY] Bug: webrtc:15210 Change-Id: I3b0e463cf3236cd9a481fbab5688643c203958da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307361 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40267}
2023-06-12 19:36:03 +08:00
"+modules/video_coding/utility/ivf_file_writer.h",
],
Add YUV to IVF video converter util Bug: webrtc:10138 Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 15:03:45 +01:00
}
Reference in New Issue Copy Permalink