If the upper limit is infinite, dont probe.
Bug: webrtc:42224658
Change-Id: Ia662cceded83969ec11ee013adb2100f983fbd13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363660
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43079}
Add field trial parameter for setting scale factor of max allocated bitrate used for deciding when to skip probing.
Currently, a factor of 2 is used in most places for max allocated bitrate but not if the field trial skip_estimate_larger_than_fraction_of_max is used.
The purpose of this new field parameter is to be able to harmonize and always use the same factor.
Bug: webrtc:42224658
Change-Id: I5e1580b9bb18ef881b819affc0b4038094e94316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363400
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43078}
This aligns to probe limits in ALR for example.
Bug: webrtc:369044000, b/369021234
Change-Id: I3823b308cf97a8b7060b35b2ac38864e75d6f983
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363301
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43074}
The reason we are doing this is because the "committers" group grants access to Code-Review, and we now have a policy that such groups must only contain direct human users. However, we still want robot accounts to be able to CQ+2, so we now have the -submit-access group that includes anyone who can CQ+2, not just committers.
Bug: chromium:356276684
Change-Id: I39c3b92110ff10f53a2825f5e12690580331364d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363480
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Auto-Submit: Joey Scarr <jsca@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43073}
Setting the standard deviation to 0 is valid and should be interpreted
as directly using the sample value at the coordinates without weighting.
This is made explicit in the documentation for the Corruption Detection
extension:
http://www.webrtc.org/experiments/rtp-hdrext/corruption-detection
Also, change stddev to std_dev in halton_frame_sampler files.
Bug: webrtc:358039777
Change-Id: Id5aa4110194f7f2b2fe9914c94304c90afd64198
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363300
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43070}
Level asymmetry is implicitly enabled for HEVC. When comparing two
codec params to see if they match, we only compare profile & tier,
similar as H.264.
Bug: chromium:41480904
Change-Id: I9e9debdf1b34f33986da9344b9fee14071b1ed60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43069}
Ensure probing is not instantiated again until after timeout if a probe has been sent to max rate.
Bug: webrtc:42224658
Change-Id: I7d0d2edcfa81b1b454ea5748962af5a2070b347c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363240
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43068}
We currently specify stream parameters to be a range for both framerate
and resolution, where preferred value is specified. The preferred value
doesn't seem to be taken into account and we end up accepting resolution
from 1x1 to MAX_INTxMAX_INT. In case the other side tries to first match
with lower resolution than requested, we will happily match it and start
streaming low quality video. We should instead request the exact stream
parameters as specified by requested capability. This capability always
come from what has been originally reported as supported so it shouldn't
happen we don't find a matching stream. This also applies to requested
video format. We previously requested mjpg for streams with resolution
higher than 640x480, but it doesn't necessarily mean the camera supports
mjpg for the requested resolution. Again, refer to requested capability
in this case as it should indicate what is supported and we know we can
request exactly the same video format. It can happen that framerate is
set to 0 as unspecified. In that case keep using a range as before, but
with more sane values.
Bug: webrtc:42225999
Change-Id: I46d8e83c636e25e12c45a462596fee1d5e59888e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362820
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Cr-Commit-Position: refs/heads/main@{#43067}
This is a reland of commit e77d75193f4f61cf90991569c5470ba5d1b78f2b.
No changes were required to the CL, downstream tests have been fixed.
Original change's description:
> Disable TLS session ticket for DTLS
>
> since it makes no sense for the WebRTC usage of DTLS and increases
> the size of the last handshake flight considerably
> Guarded by killswitch
> WebRTC-DisableTlsSessionTicketKillswitch
>
> BUG=webrtc:367181089
>
> Co-authored-by: Jody Ho <jodyho@meta.com>
> Change-Id: I4bb17bba8a17c65c8e0fefe2d8962974703feee7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362526
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43046}
Bug: webrtc:367181089
Change-Id: I4b3f813e4a0dd4d0458ee14c15c51ee6f9b84461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43066}
This reverts commit 3753c8190e3f0aca6758a5521e33f8b5d4f09ab4.
Reason for revert: Break assembling of hardware encoded h264 P frame on
weak network condition.
Original change's description:
> h264: fix first_packet_in_frame logic for multislice in a single rtp packet
>
> a frame must be (or should be) first when it contains either SPS (but not just PPS),
> is an IDR or is a slice with first_mb_in_slice == 0.
>
> Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
> into a single RTP packet which can happen with small 320x196 frames
>
> BUG=webrtc:352379280,webrtc:346608838
>
> Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42652}
Bug: webrtc:368335257
Change-Id: I07725c78be628bff71b79b8b9369677e39f5f5ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363080
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43062}
The code that restricts the maximum number of simulcast layers based on
resolution is a spec-compliance bug and doesn't make much sense: if the
app asks for 3 layers it should get 3 layers. Since the app knows the
size of the track, it could very easily ask for 1 layer when resolution
is small if that is the behavior it wanted. If the app doesn't ask to
disable layers, WebRTC shouldn't disable layers on its behalf.
This behavior makes even less sense with this "new" API since the app
is explicitly controlling the send resolution in absolute terms.
Removing this behavior in the general case is out of scope since it
would break backwards compatibility, but since `requested_resolution`
has not been exposed to the web yet and existing usage is small, this
is an opportunity to fix the compliance bug for this API.
This CL makes the last web platform test for "scaleResolutionDownTo"
pass.
Bug: chromium:363544347
Change-Id: Ic6fadf3cad69d3beec4ae03d3d031e8062382ad9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43061}
The FrameInstrumentationGenerator is responsible for generating the
instrumentation data that will be used to detect corruption. The data is
then passed to the encoder in the CodecSpecificInfo.
Bug: webrtc:358039777
Change-Id: I79d0534920b4c7fa001e1138371dfd36c13424fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362584
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43060}
In many cases, the framerate can be specified as list of possible values
and in that case, we would end up with max FPS to be set to 0 as this
case was not handled.
Bug: webrtc:42225999
Change-Id: I036af6db1da3309b1310b754504369e8fe392d09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362961
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43057}
This is a follow up for https://webrtc-review.googlesource.com/c/src/+/360680.
* Adding some missing <optional> include.
* Adding a IWYU pragma to force keeping an include.
Note that I've added the CQ bot 'iwyu_verifier' to ensure the repo stays clean. It is still work in progress and it currently needs to be triggered manually.
FYI I used these command line to run iwyu:
> for i in api/*.cc; do ./tools_webrtc/iwyu/apply-include-cleaner $i; done
> for i in api/*.h; do ./tools_webrtc/iwyu/apply-include-cleaner $i; done
Change-Id: Ie7036d08edbb6884f2b35eb9d69646757d662390
Bug: webrtc:42226242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#43054}
after cleaning up the Chromium dependency
BUG=webrtc:42225170
Change-Id: Icd3934ca51f829c55e061fc1943500435c845a8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362569
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43053}
This reverts commit e77d75193f4f61cf90991569c5470ba5d1b78f2b.
Reason for revert: Speculative rollback (breaks downstream test).
Original change's description:
> Disable TLS session ticket for DTLS
>
> since it makes no sense for the WebRTC usage of DTLS and increases
> the size of the last handshake flight considerably
> Guarded by killswitch
> WebRTC-DisableTlsSessionTicketKillswitch
>
> BUG=webrtc:367181089
>
> Co-authored-by: Jody Ho <jodyho@meta.com>
> Change-Id: I4bb17bba8a17c65c8e0fefe2d8962974703feee7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362526
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43046}
Bug: webrtc:367181089
Change-Id: I02b59232fae9f729341811042a02f7cf346d4bbe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362982
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43052}
When a value is set in RtpEncodingParameters::codec, the corresponding
payload_type will be set in the SDP a=rid: line.
a=rtpmap:96 VP8/90000
...
a=rtpmap:97 VP9/90000
...
a=rid:r0 send pt=96
a=rid:r1 send pt=97
Bug: webrtc:362277533
Change-Id: Ia9688a5fc83c53cf46621d97e87f8dd363a4d7f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43049}
Passing Environment instead of Clock into this class simplifies some plumbing for downstream consumers that need to read field trials within this class.
Bug: webrtc:362762208
Change-Id: Ia501e9f7f1d91a8115a2f71fb005dd35146db172
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362535
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43048}
since it makes no sense for the WebRTC usage of DTLS and increases
the size of the last handshake flight considerably
Guarded by killswitch
WebRTC-DisableTlsSessionTicketKillswitch
BUG=webrtc:367181089
Co-authored-by: Jody Ho <jodyho@meta.com>
Change-Id: I4bb17bba8a17c65c8e0fefe2d8962974703feee7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362526
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43046}
since clang 3.7 dates to 2015 and is obsolete.
Fuchsia has been fixed: https://issues.fuchsia.dev/issues/42051468
BUG=None
Change-Id: Id0966c3f4acfb1756eab4e2e6f0dbe52d70ecd76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362571
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43044}
This class calculates the corruption score based on the given samples from two frames.
Bug: webrtc:358039777
Change-Id: Ib036f91ec16609e827137cc35d342a2c49764737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362801
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#43043}
For key frames: increase the sequence index until the last 7 bits are
all zeroes. If this results in an overflow, wraparound to 0.
Also:
* Allow setting and getting the sequence index
* Allow getting LayerId
Bug: webrtc:358039777
Change-Id: Ibe16689a3d1eb5706d4fce5c9220770046f26896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362540
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43042}
We need to have imported proto as proto_data_sources in BUILD.gn to
run the action remotely without workaround config in siso.
Bug: b/366137880
Change-Id: I053774f00b761520a8a85154e386da3edb8f39b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362680
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/main@{#43040}
which is no longer used. Also the blink::WebRTCKeyType it refers
to no longer exists either
BUG=None
Change-Id: I8236ed906b5712d11173dfcf181f556b1ff597f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362387
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43038}
The `requested_resolution` API must not change aspect ratio, example:
- Frame is 60x30
- Requested is 30x30
- We expect 30x15 (not 30x30!) as to maintain aspect ratio.
This bug was previously fixed by making VideoAdapter unaware of the
requested resolution behind a flag: this seemed OK since the
VideoStreamEncoder ultimately decides the resolution, whether or not
the incoming frame is adapted.
But this is not desired for some non-Chrome use cases. This CL attempts
to make both Chrome and non-Chrome use cases happy by implementing the
aspect ratio preserving restriction inside VideoAdapter too.
This allows us to get rid of the "use_standard_requested_resolution"
flag and change the "VideoStreamEncoderResolutionTest" TEST_P to
TEST_F.
Bug: webrtc:366067962, webrtc:366284861
Change-Id: I1dfd10963274c5fdfd18d0f4443b2f209d2e9a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362720
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43037}
Use the DTX state from inside the Opus encoder instead of trying to
mimic the logic outside.
Bug: None
Change-Id: I852044fee261a5b7f9255c557a27adfd0b1701bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43034}
This adds payload types to the codecs at the time when offer
is being generated, if they are unassigned at that point.
Bug: webrtc:360058654
Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43033}
They are now passed as part of the Environment
Bug: webrtc:362762208
Change-Id: I02868e9f41533a546f62fe30fdc6f3a7708eb346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362084
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43032}
This will help to reduce redundant ScalabilityMode to temporal layer
count mapping in blink.
Bug: chromium:40763991
Change-Id: Ida3e6abb91383e27465eb1b697ad9431935cf9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362486
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43031}
The recently added tests resulted in some .log file on some bots being
too long:
video_engine_tests_exe-VideoStreamEncoderStandardOrLegacyRequestedResolutionTest_VideoStreamEncoderStandardOrLegacyRequestedResolutionTest_RequestedResolutionInWrongAspectRatioAndSourceIsAdapting_0-1.log
This CL makes the test names significantly shorter.
# Trivial and believed to fix purple bots, let's land ASAP
NOTRY=True
Bug: webrtc:367066321
Change-Id: I831911947af9d5639d1edb559470f1c9ae702d6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362721
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43030}
