This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1.
Reason for revert: breaks a downstream project
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
Note: this CL has to leave behind one part of iSAC, which is its VAD
currently used by AGC1 in APM. The target visibility has been
restricted and the VAD will be removed together with AGC1 when the
time comes.
Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
Bug: webrtc:14450
Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38652}
The DCHECK crashes debug builds running some applications such as Webex.
Bug: None
Change-Id: I0061286c4c1d04964678a00014896f1fccd4685d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276460
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38644}
Address perkj's comments left in
https://webrtc-review.googlesource.com/c/src/+/283420. I was a bit
trigger-happy with the submit button.
Bug: chromium:1354491
Change-Id: Ifd052f75af3763b0b52807c31ea790e3efee921d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38638}
I've been working with SizeBench (binary size analysis tool) and it
reported that 39 virtual functions were not overridden. Removed
virtual keyword from each. SizeBench estimated waste 2.1kb. Change
made chrome.dll 5.3kb smaller. Since these 39 virtual functions
are never overridden, they are wasteful.
Note: These are the savings for Windows, relocation savings are probably larger on other platforms.
GN args for builds:
use_goma=true
is_debug=false
target_cpu="x64"
use_lld=false
fatal_linker_warnings=false
symbol_level=2
dcheck_always_on = false
pe_summarize analysis pre-change -> change:
Size of out\Default\chrome.dll is 187.205120 MB
Size of out\MediaContentDescription\chrome.dll is 187.199488 MB
Memory size change from out\Default\chrome.dll to
out\MediaContentDescription\chrome.dll
.text: -2624 bytes change
.rdata: -1984 bytes change
.pdata: -48 bytes change
.reloc: -644 bytes change
Total change: -5300 bytes
Bug: chromium:1371503
Change-Id: Ib33829fada54abdf8fed33ec96f11a03ce6fcb68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281442
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivan Rosales <rosalesi@google.com>
Cr-Commit-Position: refs/heads/main@{#38630}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.
When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.
NOTRY=True
Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
The file was renamed, see
https://groups.google.com/u/1/g/discuss-webrtc/c/ZQiP4f_bpw4
Bug: webrtc:14180
Change-Id: Ia76c85ba7d9da6b3a93d0a67a4b6a5187e07e230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283084
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38616}
As to not break downstream projects, the old name api/stats_types.h is
kept around to help include api/legacy_stats_types.h. We can delete this
in a follow-up.
NOTRY=True
Bug: webrtc:14180
Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38610}
This CL finalizes the Metronome refactor undertaken in
crbug.com/1381982 and enables it again in call.cc.
Fixed: chromium:1381982
Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38605}
The Chromium implementation unfortunately has a rare deadlock.
Rather than patching that up, we're changing the metronome
implementation to be able to use a single-threaded environment
instead.
The metronome functionality is disabled in VideoReceiveStream2
construction inside call.cc.
The new design does not have listener registration or
deresigstration and instead accepts and invokes callbacks, on
the same sequence that requested the callback. This allows
the clients to use features such as WeakPtrFactories or
ScopedThreadSafety for cancellation.
The CL will be followed up with cleanup CLs that removes
registration APIs once downstream consumers have adapted.
Bug: chromium:1381982
Change-Id: I43732d1971e2276c39b431a04365cd2fc3c55c25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38582}
ChannelManager has been deleted, these declaration should also be deleted.
Bug: webrtc:13931
Change-Id: I2739a0424f61d6e659cb694a3f51bb6b90911cf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282520
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38579}
As the exposure of power efficient stats to JavaScript are limited as
to reduce the fingerprinting surface to getStats, a new RTCStatsMember
derivation, RTCLimitedStatsMember, was added in this change. This sets
the exposure criteria of the stat on the type, which keeps the size of
the RTCStatsMember class the same and allows for extension in the future
for new types of stat restrictions.
Bug: webrtc:14483
Change-Id: Ib0303050a112441ba2416fd5f004dd8be26b47ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279021
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38576}
and do the resolution of rids to layers. This has no effect yet
since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer.
BUG=chromium:1354101
Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38565}
replacing
if (optional) { ...}
with the more explicit
if (optional.has_value()) { ... }
No functional changes.
BUG=None
Change-Id: I005fd3df307880b07cfda0cbe435efb0e0717a88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281362
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38544}
This is a reland of commit b395f5bd5c5be7a3b54fc85ca482759a56a0cd6e
Downstream project has been fixed.
Original change's description:
> move relay server priority assignment to port_allocator
>
> which knows more about the internals of ICE.
> Remove the relay server config priority field which was used to
> specify the relative priority of TURN servers. This is now handled
> internally by CreateRelayPortArgs without being exposed.
>
> Also rename BasicPortAllocator::AddTurnServer to
> BasicPortAllocator::AddTurnServerForTesting since it is a test-only
> method.
>
> BUG=webrtc:13195,webrtc:14539
>
> Change-Id: Id36cbf0187b7a84d1a9b53860f31994f3c7589f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280224
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38520}
Bug: webrtc:13195,webrtc:14539
Change-Id: I617b611de97e4013c7286e3345073ca7589065c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281662
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38543}
out of the DTLS transport creation and into the ICE transport
creation.
BUG=None
Change-Id: I9f1a74bbabde842234e5ae7cc64aa9632ab2ba78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38541}
These have all been moved to "inbound-rtp" and now that upstream
projects have migrated we can delete the old location.
Unblocks https://crbug.com/webrtc/14175
Bug: webrtc:14521, webrtc:14524
Change-Id: Ia2bfa399d62304cc0ead0e65c340dfad20acc530
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38532}
This reverts commit b395f5bd5c5be7a3b54fc85ca482759a56a0cd6e.
Reason for revert: Breaks downstream project. Jonas will help to reland this CL.
Original change's description:
> move relay server priority assignment to port_allocator
>
> which knows more about the internals of ICE.
> Remove the relay server config priority field which was used to
> specify the relative priority of TURN servers. This is now handled
> internally by CreateRelayPortArgs without being exposed.
>
> Also rename BasicPortAllocator::AddTurnServer to
> BasicPortAllocator::AddTurnServerForTesting since it is a test-only
> method.
>
> BUG=webrtc:13195,webrtc:14539
>
> Change-Id: Id36cbf0187b7a84d1a9b53860f31994f3c7589f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280224
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38520}
Bug: webrtc:13195,webrtc:14539
Change-Id: I7ca087a272793908f003cea6c32efe6214e54028
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281340
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38524}
which knows more about the internals of ICE.
Remove the relay server config priority field which was used to
specify the relative priority of TURN servers. This is now handled
internally by CreateRelayPortArgs without being exposed.
Also rename BasicPortAllocator::AddTurnServer to
BasicPortAllocator::AddTurnServerForTesting since it is a test-only
method.
BUG=webrtc:13195,webrtc:14539
Change-Id: Id36cbf0187b7a84d1a9b53860f31994f3c7589f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280224
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38520}
split from the actual change for easier review and cherry-picking.
BUG=webrtc:14605
Change-Id: I7dbaf8e1f4a03f35a5d8c4da1a2784b00589bfc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280680
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38519}
These metrics were not only non-standard, but residing in the
non-standard "track" stats object that we want to delete. As per
https://github.com/w3c/webrtc-stats/issues/695#issuecomment-1259611462
these metrics are no longer needed because we already have
inbound-rtp.totalInterFrameDelay/totalSquaredInterFrameDelay which is
basically the same thing.
// mac_rel infra failures are unrelated
NOTRY=True
Bug: webrtc:14522
Change-Id: I565da42514a93f15532ba8357dd006547a5296ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38509}
the foundation attribute is currently calculated as
CRC32(baseaddress, protocol, relayprotocol)
which is a way to satisfy the requirements from
https://www.rfc-editor.org/rfc/rfc5245#section-4.1.1.3
However, this leaks the base address which defeats the
MDNS obfuscation described in
https://datatracker.ietf.org/doc/draft-ietf-mmusic-mdns-ice-candidates/
since the CRC32 can be reversed using a table lookup as shown in
https://github.com/niespodd/webrtc-local-ip-leak/
To defeat that lookup, "seed" the CRC32 with the ICE tie-breaker which is a randomly picked unsigned 64 bit integer described in
https://www.rfc-editor.org/rfc/rfc5245#section-5.2
The tie-breaker is not known to Javascript and adding it scopes the foundation within the peer connection as described in section 4.1.1.3
To manually test (preferably with a DCHECK for IceTiebreaker() in ComputeFoundation)
- gather candidates twice on https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/ and observe that the foundations are not the same after this change
- create two RTCPeerConnections with {iceCandidatePoolSize: 1}, create a datachannel, call setLocalDescription, inspect the candidates and observe that the foundations are not the same after this change.
Unit test changes have been split into a separate CL for easier integration.
BUG=webrtc:14605
Change-Id: I6bbad1635b48997b00ae74d251ae357bf8afd12f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38485}
This information is now readily available. Let's expose it.
In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.
Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
This metric was always supposed to be the spec's answer to
googBucketDelay, and is defined as "The total number of seconds that
packets have spent buffered locally before being transmitted onto the
network." But our implementation measured the time between capture and
send, including encode time. This is incorrect and yields a much larger
value than expected.
This CL updated the metric to do what the spec says. Implementation-wise
we measure the time between pushing and popping each packet from the
queue (in modules/pacing/prioritized_packet_queue.cc).
The spec says to increment the delay counter at the same time as we
increment the packet counter in order for the app to be able to do
"delta totalPacketSendDelay / delta packetSent". For this reason,
`total_packet_delay` is added to RtpPacketCounter. (Previously, the
two counters were incremented on different threads and observers.)
Running Google Meet on a good network, I could observe a 2-3 ms average
send delay per packet with this implementation compared to 20-30 ms
with the old implementation. See b/137014977#comment170 for comparison
with googBucketDelay which is a little bit different by design -
totalPacketSendDelay is clearly better than googBucketDelay.
Since none of this depend on the media kind, we can wire up this metric
for audio as well in a follow-up:
https://webrtc-review.googlesource.com/c/src/+/280523
Bug: webrtc:14593
Change-Id: If8fcd82fee74030d0923ee5df2c2aea2264600d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280443
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38480}
defined in
https://w3c.github.io/webrtc-encoded-transform/#rtcrtpsender-extension
Note: this does not implement the "rid(s)" parameter which will be done in a future CL.
VP8 still synchronizes keyframes on all layers even when asked for ones on individual layers while H264 (when implemented as three different encoders in SimulcastEncoderAdapter) can actually utilize this.
This does not change the behavior when receiving a RTCP PLI for a particular layer.
BUG=chromium:1354101
Change-Id: Ic8b14d155242e32c9aeafa55fe6652f346ac76b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274169
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38472}
This is exposing something that is already exposed in the legacy
getStats() API and is only available if the "video-timing" header
extension is used. Adding this metric here should unblock legacy
getStats() API deprecation. The follow-up to unship or standardize this
metric is tracked by https://crbug.com/webrtc/14586.
Bug: webrtc:14587
Change-Id: Ic3d45b0558d7caf4be2856a4cd95b88db312f85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38444}
This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats
There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.
In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.
# Unrelated infra failures
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
This patch adds variant of PeerConnectionInterface::AddTrack
that takes an initial_send_encodings.
This allows for setting/modifying encoding parameters before sdp
negotiation is performed/complete (e.g requested_resolution).
This is already available if using RtpTransciverInit and AddTransceiver,
but was not added to AddTrack because of concerns that it complicated matching with existing transceivers. This CL sidesteps that by never matching to a preexisting transceiver if initial_send_encodings are specified.
Note:
1) The patch adds a new method rather than an extra (e.g optional)
argument to existing AddTrack. This is to avoid problems with downstream mocks.
2) chromium "problems" was fixed in https://chromium-review.googlesource.com/c/chromium/src/+/3952684 and https://chromium-review.googlesource.com/c/chromium/src/+/3956060
Bug: webrtc:14451
Change-Id: I19b5a03872730280fbf868ca5d3a2f46443359f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38437}
Typically, remote candidates come from signalling and are deserialized
into C++ objects. The network_type field of these candidates is
always ADAPTER_TYPE_UNKNOWN.
However, in tests it is common to pass SDP and remote candidates as C++
objects. In this case, the network_type property of remote candidates
is preserved, so DCHECK might be triggered when GetStats is called.
Clearing fields that are not suitable as remote candidates fixes
this issue.
Bug: None
Change-Id: Ida01b0224bce5cf3e87bcad1ddaca35c9f4fffe7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38436}
ScalabilityMode should be validated against the currently
allowed codecs or the currently used codec.
Bug: webrtc:11607
Change-Id: Id2e6cbfad4f089de450150e1203657ed316e2f29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277403
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38433}
which have shown that it is not easily possible to restrict
the pool size to 1 and combine this with max-bundle
BUG=webrtc:12383,chromium:1328218
Change-Id: I3a7ae4a263238c1b5faa079c3cbdaf84d1b40cbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38396}
This is a reland of commit c4b0bde7f2daabc4e0667fb73d096d1cf0819226
which changes the name of the new method and adds a deprecated
backward compatible variant with the old name.
Original change's description:
> ice server parsing: return RTCError with more details
>
> surfacing those errors to the API (without specific details) instead of just the logging.
>
> BUG=webrtc:14539
>
> Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38356}
Bug: webrtc:14539
Change-Id: I0a5482e123f25867582d62101b31ed207b95ea1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278962
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38364}
This reverts commit c4b0bde7f2daabc4e0667fb73d096d1cf0819226.
Reason for revert: Breaks downstream tests.
Basically, ParseIceServers() and other functions have changed
the return type, and this breaks tests at compile time.
Is it possible to reland with backwards compatible versions that return
the previous type? Then they can be removed afterwards.
Original change's description:
> ice server parsing: return RTCError with more details
>
> surfacing those errors to the API (without specific details) instead of just the logging.
>
> BUG=webrtc:14539
>
> Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38356}
Bug: webrtc:14539
Change-Id: I4df936ff865f87759936c5d0425478fe51051dc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278960
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38359}
surfacing those errors to the API (without specific details) instead of just the logging.
BUG=webrtc:14539
Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38356}
which have shown that 32 is a reasonably safe limit and informed
https://github.com/w3c/webrtc-pc/pull/2679/
BUG=webrtc:13265,chromium:1360224
Change-Id: I63155653862e4c12720b8655c51ed5f3bdc742f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277804
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38339}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.
To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.
Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
These metrics were recently standardized. Part of the standardization
effort was to move them from obsolete "track" stats (on track for
deprecation and removal: https://crbug.com/webrtc/14175) into the
"inbound-rtp" stats which are not deprecated.
To ease transition for downstream projects, the metrics are temporarily
duplicated in both the old and new locations. In a follow-up CL, they
will be deleted from "track".
Bug: webrtc:14521
Change-Id: I0d9036472607a8c717ec823a458a79a49ddb80c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38308}
