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}
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}
This will allow us to optimize the internal buffers of
webrtc::VideoFrame for the resolution(s) that we actually want to
encode.
Bug: webrtc:12469, chromium:1157072
Change-Id: If378b52b5e35aa9a9800c1f7dfe189437ce43253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33342}
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
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@{#33325}
Currently the echo canceller reference signal is high-pass filtered to
avoid the need of modeling the capture-side high-pass filter as part of
the echo path.
This can lead to the lowest frequency bins of the linear filter
diverging as there is little low-frequency content available for
training. Over time the filter can output an increasing amount of
low-frequency power, which in turn affects the filter's ability to
adapt properly.
Disabling the high-pass filtering of the echo canceller reference solves
this issue, resulting in improved filter convergence.
Bug: webrtc:12265
Change-Id: Ic526a4b1b73e1808cfcd96a8cdee801b96a27671
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208288
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33322}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
This make it easier to create parameters from a single endpoint ptr.
Bug: None
Change-Id: Id64757353505a21c7731655e1b7a3178fa2e5ef8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207425
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33263}
Audio interruption metric is not implemented for codecs doing their own PLC.
R=ivoc@webrtc.org, jakobi@webrtc.org
Bug: b/177523033 webrtc:12456
Change-Id: I0aca6fa5c0ff617e76ee1e4ed8d95703c7097223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206561
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@google.com>
Cr-Commit-Position: refs/heads/master@{#33229}
