SdpVideoFormat is used to configure video encoder and decoders.
This CL adds support for comparing two SdpVideoFormat objects
to determine if they specify the same video codec.
This functionality previously only existed in media/base/codec.h
which made the code sensitive to circular dependencies. Once
downstream projects stop using cricket::IsSameCodec, this code
can be removed.
Bug: chromium:1187565
Change-Id: I242069aa6af07917637384c80ee4820887defc7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213427
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33794}
On a 32bit system, this reduces the allocation size of the flag
down from 12 bytes to 8, and removes the need for a vtable (the extra
4 bytes are the vtable pointer).
The downside is that this change makes the binary layout of the
flag, less compatible with RefCountedObject<> based reference counting
objects and thus we don't immediately get the benefits of identical
COMDAT folding and subsequently there's a slight binary size increase.
With wider use, the binary size benefits will come.
Bug: none
Change-Id: I04129771790a3258d6accaf0ab1258b7a798a55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33793}
The ERLE is used to estimate residual echo for echo suppression. The
ERLE is reduced during far-end offset to avoid echo leakage. When there
is a strong near-end present this can cause unnecessary transparency loss.
This change adds an ERLE estimation that does not compensate for onsets and
uses it for residual echo estimation when the suppressor considers the near-end to be dominant.
Bug: webrtc:12686
Change-Id: Ida78eeacf1f95c6e62403f86ba3f2ff055898a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215323
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33786}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
Since there is only a single type of DataChannel now, the enum was only used
when data channels were disabled at the PC API. That option has been
deprecated 4 years ago, it's now time to remove it.
Bug: webrtc:6625
Change-Id: I9e4ada1756da186e9639dd0fbf0249c55ea0b6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33778}
SequenceChecker needs to be prefixed with & in RTC_DCHECK_RUN_ON;
all examples except the first one were showing this.
Bug: none
Change-Id: I90468689675319f9df67eb04a5d4cc0767ffb7a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215582
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33771}
This denies the ability to request RTP data channels to callers.
Later CLs will rip out the actual code for creating these channels.
Bug: chromium:928706
Change-Id: Ibb54197f192f567984a348f1539c26be120903f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33740}
This reverts commit a743303211.
Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?
Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
> non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
> is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}
TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
Since we want most users to use the PeerConnection API, this is the
part that we should document.
If we want people to use other APIs, we can add to the file.
Bug: webrtc:12674
Change-Id: Icf14f218cf51c640e6f846f10b49dff84106dc21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215066
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33707}
This is the result of compiling Chromium with
Wtautological-unsigned-zero-compare. For more details, see:
https://chromium-review.googlesource.com/c/chromium/src/+/2802412
Change-Id: I05cec6ae5738036a56beadeaa1dde5189edf0137
Bug: chromium:1195670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33689}
This change adds support for emitting encoded frames
for recording when the decoder can't easily read out
encoded width and height as is the case for AV1 streams,
in which case the information is buried in OBUs. Downstream
project relies on resolution information being present for key
frames. With the change, VideoReceiveStream2 infers the
resolution from decoded frames, and supplies it in the
RecordableEncodedFrames.
Bug: chromium:1191972
Change-Id: I07beda6526206c80a732976e8e19d3581489b8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33662}
This reverts commit 5a40b37105.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
Before, the calls went through the signaling thread and
blocked while the operation completed on the worker.
Bug: webrtc:12601
Change-Id: I58991fa98a55d0fa9304a68bd85bb269f1f123d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212619
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33615}
Packets, chunks, parameters and error causes - the SCTP entities
that are sent on the wire - are buffers with fields that are stored
in big endian and that generally consist of a fixed header size, and
a variable sized part, that can e.g. be encoded sub-fields or
serialized strings.
The BoundedByteReader and BoundedByteWriter utilities make it easy
to read those fields with as much aid from the compiler as possible,
by having compile-time assertions that fields are not accessed
outside the buffer's span.
There are some byte reading functionality already in modules/rtp_rtcp,
but that module would be a bit unfortunate to depend on, and doesn't
have the compile time bounds checking that is the biggest feature of
this abstraction of an rtc::ArrayView.
Bug: webrtc:12614
Change-Id: I9fc641aff22221018dda9add4e2c44853c0f64f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212967
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33597}
This API should allow existing factories to be used unmodified, but
offers a new API that documents ownership better and does not use
sigslot.
Bug: webrtc:12598
Change-Id: I0f68371059cd4a18ab07b87fc0e7526dcc0ac669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212609
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33553}
Deleting obsolete stats. Spec: https://www.w3.org/TR/webrtc-stats/
1. RTCInbound/OutboundRtpStats.isRemote: No longer useful with remote stream stats
2. RTCIceCandidateStats.deleted: This field was obsoleted because if the ICE candidate is deleted it no longer appears in getStats()
I also marked as many other obsoleted stats possible according to spec. I am not as confident to delete them but feel free to comment to let me know if anything is off / can be deleted.
Bug: webrtc:12583
Change-Id: I688d0076270f85caa86256349753e5f0e0a44931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33549}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
This avoids falling back on the VideoFrameBuffer::CropAndScale default
implementation which performs ToI420. This has two major benefits:
1. We save CPU by not converting to I420 for NV12 frames.
2. We make is possible for simulcast encoders to use Scale() and be
able to trust that the scaled simulcast layers have the same pixel
format as the top layer, which is required by libvpx.
In order to invoke NV12Buffer::CropAndScaleFrom() without introducing a
circular dependency, nv12_buffer.[h/cc] is moved to the "video_frame"
build target.
Bug: webrtc:12595, webrtc:12469
Change-Id: I81aac5c6b3e81c49f32a7be6dc2640e6b40f7692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33521}
0 is not valid per https://tools.ietf.org/html/rfc5245#section-15.1 which says
<component-id>: is a positive integer between 1 and 256
This is part of the RTCIceTransport extension API which is probably going away.
BUG=chromium:1044875
Change-Id: I56d8dec79d3191e084f4a25a2c0a4d0b67afde74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33515}
This reverts commit 31c5c9da35.
Reason for revert: made QP parser thread-safe https://webrtc.googlesource.com/src/+/0e42cf703bd111fde235d06d08b02d3a7b02c727
Original change's description:
> Revert "Reland "Enable quality scaling when allowed""
>
> This reverts commit 0021fe7793.
>
> Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
>
> Original change's description:
> > Reland "Enable quality scaling when allowed"
> >
> > This reverts commit eb449a979b.
> >
> > Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
> >
> > Original change's description:
> > Before this CL quality scaling was conditioned on scaling settings
> > provided by encoder. That should not be a requirement since encoder
> > may not be aware of quality scaling which is a WebRTC feature. In M90
> > chromium HW encoders do not provide scaling settings (chromium:1179020).
> > The default scaling settings provided by these encoders are not correct
> > (b/181537172).
> >
> > This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> > is set to true in singlecast with normal video feed (not screen sharing)
> > mode. If quality scaling is allowed it is enabled no matter whether
> > scaling settings are present in encoder info or not. Setting from
> > QualityScalingExperiment are used in case if not provided by encoder.
> >
> > Bug: chromium:1179020
> > Bug: webrtc:12511
> > Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33438}
>
> TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com
>
> Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1179020
> Bug: webrtc:12511
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33439}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I3a31e1c1fdf7da93226f8c1e0a96b43fe0b786df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212026
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33481}
Use PendingTaskSafetyFlag for safe Stop. Followup to
https://webrtc-review.googlesource.com/c/src/+/209181.
Also fix rtc::scoped_refptr to work with RTC_PT_GUARDED_BY.
Bug: webrtc:12339
Change-Id: Ic0e3ecb17049f1a0e6af887ba5f97a5b48a32d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211351
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33447}
This reverts commit 0021fe7793.
Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit eb449a979b.
>
> Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33438}
TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com
Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1179020
Bug: webrtc:12511
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33439}
This reverts commit eb449a979b.
Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33438}
Spec: https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
According to the spec, |RTCReceivedRtpStreamStats| is the base class for |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. This structure isn't visible in JavaScript but it's important to bring it up to spec for the C++ part. This CL adds the barebone |RTCReceivedRtpStreamStats| with a bunch of TODOs for later migrations.
This commit makes the minimum |RTCReceivedRtpStreamStats| and rebase |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| to use the new class as the parent class.
This commit also moves |jitter| and |packets_lost| to |RTCReceivedRtpStreamStats|, from |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. Moving these two first because they are the two that exist in both subclasses for now.
Bug: webrtc:12532
Change-Id: I0ec74fd241f16c1e1a6498b6baa621ca0489f279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210340
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33435}
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33404}
This reverts commit 83be84bb74.
Reason for revert: Suspect of crbug.com/1185276
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33385}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33393}
This reverts commit 609b524dd3.
Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33385}
This reverts commit 752cbaba90.
Reason for revert: The test VideoStreamEncoderTest.QualityScalingAllowed_QualityScalingEnabled seems to fail on iOS.
Original change's description:
> Enable quality scaling when allowed
>
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020, webrtc:12511
> Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33373}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Icabf2d9a034d359f79491f2c37f1044f17a7445d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209641
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33381}
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33373}
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.
Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33366}
Tests:
./out/Default/peerconnection_unittests
Manually tested with Chromium to see the data populated
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33361}
This is a reland of a37f2bd942
Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}
Bug: none
Change-Id: If46a6679ac0fc947797dd7be87626ef7702faca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208845
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33349}
This reverts commit a37f2bd942.
Reason for revert: Breaks compile step (e.g. https://ci.chromium.org/ui/p/webrtc/builders/ci/Android64%20Builder%20x64%20(dbg)/19773/overview).
Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}
TBR=hta@webrtc.org
Bug: none
Change-Id: I2014faab3392f445f56edd9e833d00000ebc5ca3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33347}
