This CL cleans up all local conversions, in favor of the common helper
function.
Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
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}
under the same conditions as video_replay.
Drive-by: fix typos
BUG=webrtc:15210
Change-Id: I6d288b2f7c8e2101192556eada6b28c82bfabf2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308723
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40355}
As before, events are processed primarily in timestamp order.
This CL adds a heuristic to break ties for events with the same timestamp.
- Roughly speaking, configs and connectivity events are processed first, followed by incoming packets, then BWE updates, then other (general) events and finally outgoing packets and ALR events.
- Among RTP packets, transport sequence number is used to break ties.
- The insertion order (into the EventProcessor) is used as a last resort.
Bug: b/282153758
Change-Id: I914e4500ca63e1a8754766d1833a7b32f6a38176
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308140
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40318}
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}
The same observer implementation was being used for both client and
server but the role is different (sender vs receiver), so I split
the functionality up into two separate classes.
Bug: webrtc:11547
Change-Id: Ia60ab96fb86b4ff61fa7bff5f30d59b6fe0f9746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300742
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39886}
This can happen when there are audio SSRCs in the event log without any
associated events.
Issue was introduced in
https://webrtc-review.googlesource.com/c/src/+/300300
Bug: None
Change-Id: Ib0e009095bf67633812d937aa5a9e65e2cd8958a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300743
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39806}
Remove duplicate implementions and complex inheritance.
Slight change to the event log visualizer NetEq simulations to only
include time after the first packet has been received.
Bug: None
Change-Id: I8a7bd3d4d2b601fc134292554476020f9b3eee92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300300
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39773}
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}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
An RTP dump may or may not include the payload of the recorded RTP packets. When the payload is not present packets should still be created with their original packet length.
Bug: webrtc:14801
Change-Id: Ice74cb5f7d370aaefac5f370445ffd3f2fc5924c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289920
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38979}
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 is to simplify implementation of new feature flags.
- Move helper functions to anonymous namespace.
- Add members to avoid passing everything by function paramaters.
Bug: webrtc:14508
Change-Id: I0a4958645a4eb76515f28d8ce868a66be6748919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38274}
Early return will cause `call` to be destroyed outside the worker thread, which gives confusing error messages when all you did was type the wrong path to the input file :)
Bug: webrtc:14508
Change-Id: I029910d8da4bc7b08dafd02cb5ebf88d9c7afa59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38254}
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).
The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.
Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
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}
Making it clear that the field is used to store the applied input
volume and not the recommended input volume.
Bug: webrtc:7494, b/241923537
Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38051}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
This reverts commit 83db78e854.
Reason for revert: Some tests in Chromium's blink no longer compile because of the change in the signature of the CreateDefaultTaskQueueFactory() function.
Original change's description:
> Add TaskQueueStdlib experiment.
>
> Bug: webrtc:14389
> Change-Id: I23c6e0ae675748ec35a99c334104dd2654995a33
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265802
> Commit-Queue: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37888}
Bug: webrtc:14389
Change-Id: If3e63d6b4ab9e838dc5020b88076a73fd29916e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272920
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37902}
In migrating rtc::Event to use TimeDelta instead of int,
rtc::Event::kForever will have to become something else.
This change removes dependencies on that kForever is int.
Bug: webrtc:14366
Change-Id: Ic36057dda95513349e7ae60204e7271ff1f58825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271288
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37795}
Starting from Go 1.17, modules are the default and cannot be disabled.
This requires a change in the GitHub repositury [1].
As a stopgap solution, this CL moves Go back to 1.16 and disables
modules [2].
[1] - https://github.com/webrtc/apprtc/tree/master/src/collider
[2] - https://go.dev/blog/go116-module-changes
No-Presubmit: True
Bug: webrtc:14342
Change-Id: Idd03639588bc03497a78f0cef350daebf3b2f1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271481
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37760}
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}
Instead of creating a TaskQueue from packet_sender, create a rtc::Thread
in test_controller so that test_controller instantiates a SocketServer,
eliminating the use of rtc::Thread::socketserver().
Also did various cleanups, such as adding threading annotations, and
ensuring that all network operations are done in dedicated threads.
Bug: webrtc:13145
Test: Unittest, and manually verified using Android clients and Linux servers
Change-Id: I05ebe5e29bd80f14a193c9ee8b0bf63a1b6b94d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263321
Commit-Queue: Daniel.l Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37411}
which is useful for validating flexfec recovers frames correctly.
This can be tested by passing a keyframe covered by flexfec along
with the fec packets and removing one packet from the frame.
BUG=None
Change-Id: Icd73eca138f62b9387bf850a6efbd7db03b4b569
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261956
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37392}
which is already supported in modules/video_coding/utility/ivf_file_reader.cc
BUG=None
Change-Id: I7b00659dc460d372312dff2eb53837a321ab16af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37390}
and fix a typo
BUG=None
Change-Id: Ie286c52cbdfbc0269d92503c46b805bfdf5bb556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264151
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37042}
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}
- Filter out very old packets (to ensure that the estimate doesn't drop to zero if sending is paused and later resumed).
- Discard packets older than previously discarded packets (to avoid the estimate dropping after deep reordering.)
- Add tests cases for high loss, deep reordering and paused/resumed streams to unittest.
- Remove some field trial settings that have very minor effect and rename some of the others.
- Change analyzer.cc to only draw data points if the estimators have valid estimates.
Bug: webrtc:13402
Change-Id: I47ead8aa4454cced5134d10895ca061d2c3e32f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236347
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36849}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
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}
