Parses log, calls analyzer and populates output.
Currently only outputs two charts. Chart selection to be added in a followup.
Bug: None
Change-Id: I960cff15a5935a638a5d979a71230ad598083596
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41000}
kDefaultQpMax=56 was defined in multiple places. Move it to media_constants and split it into two: VPx/AV1 and H26x values. H26x value is set to 51 which is the max bitstream QP value for H264/5.
This CL is expected to be a no-op because:
1. VideoCodec::qpMax value has not changed for VP8/9 and AV1.
2. VideoCodec::qpMax is currently not used by OpenH264 wrapper (wiring it up is out-of-scope of this CL).
3. Previous default qpMax=56 exceeded the max value for H26x (=51). External HW H26x encoders likely clamped it and used 51.
Bug: webrtc:14852
Change-Id: I1d795e695dac5c78e86ed829b24281e61066f668
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324282
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40997}
This is currently a stub. The analysis will be added in a followup CL.
No-Try: True
Bug: None
Change-Id: Ief381d0c30ec29a0ef170523d31f1f902d0e6b09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40989}
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}
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}
Receivers no longer need to set extensions in the configuration. That field will be removed in a follow up.
Tested with:
video_loopback --rtp_dump_name="./my.rtpdump" --duration=10
video_replay --input_file=./my.rtpdump
Bug: webrtc:14795
Change-Id: I952cd487cb2f3be8be01a90f6a2312f1fef5d93e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290995
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39136}
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.
Test failures seem unrelated, so using No-Try.
No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
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}
This allow to remove the testonly from the rtc_event_log_visualizer
binary and the implicit dependency on the path of the default
conversational speech file.
The binary size of event_log_visualizer passes from 2.1 MB to 4.0 MB.
Bug: b/237526033
Change-Id: I71cf647f039f26f30c792c49c752cff5c5b329a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267663
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37453}
This CL duplicates a few lines of utility code from
//modules/audio_processing:audioproc_test_utils (which contains more
testonly things) and allows the possibility to remove testonly from
the unpack_aecdump tool.
Bug: b/237526033
Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37437}
To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36932}
Create a server using:
./data_channel_benchmark --server --port 12345
Start the flow of data from the server to a client using:
./data_channel_benchmark --port 12345 --transfer_size 100
The throughput is reported on the server console.
The negotiation does not require a 3rd party server and is done over a
gRPC transport. No TURN server is configured, so both peers need to be
reachable using STUN only.
Bug: webrtc:13288
Change-Id: Iac9a96cf390ab465ea45a46bf0b40950c56dfceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36206}
This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.
Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
This reverts commit cd5127b11e.
Reason for revert: Fuzzer build problems fixed.
Original change's description:
> Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
>
> This reverts commit dfe19719e5.
>
> Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.
>
> Original change's description:
> > Enable use of rtc::SystemTimeNanos() provided by Chromium
> >
> > This is the third CL out of three to enable overriding
> > of the function SystemTimeNanos() in rtc_base/system_time.cc
> >
> > When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> > function provided by Chromium will be used. This is controlled
> > by the build argument rtc_exclude_system_time which directly
> > maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
> >
> > By doing this we are making sure that the WebRTC and Chromium
> > clocks are the same.
> >
> > Bug: chromium:516700
> > Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33337}
>
> TBR=kron@webrtc.org
>
> Bug: chromium:516700
> Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33340}
Bug: chromium:516700
Change-Id: I4cd68bac1cc4befdb46351f5d6fb2cf1ef5c3062
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208742
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33341}
