This is a reland of f79bfc65e52a35d27cf0db2d212e94043fb44da3
the tests that have blocked the roll have been marked as allowed to fail.
Original change's description:
> peerconnection: prefer spec names for signaling state
>
> Map the internal state names to the spec ones defined in
> https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum
> instead of exposing them. This only affects the (not specified)
> error strings.
>
> Bug: None
> Change-Id: Ib0b35bb3106b1688e8386f6fdd0b8c7fdebaf1dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178390
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#31591}
Bug: chromium:1101699
Change-Id: Ia21cec9e76fbaa4df2fa5a80409a7c80fedc4faa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178562
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31914}
Instead it should be ensured that it is started with the correct config.
This removes confusion regarding a resource state. If the resource
check is stopped then the adaptations for that resource should be
removed, and there is no way to determine that if we have one method for
stop for both reconfigure and shutdown.
Bug: webrtc:11843
Change-Id: I491f2fd1f4f803a4610124c7b0026ad75ab4a9cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181368
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31913}
Deletes all webrtc usage of this member. Next step is to delete
any downstream references, and when that's done, the member can be
deleted.
Bug: None
Change-Id: I3f3a94a063dccf56468a1069653efd3809875b01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181201
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31911}
When user touches switch camera it takes a while for camera to switch.
So disabled switch camera button on touch and enabled it back when
switch camera is done. It also gives visual feedback to user.
Did same change for route change button as well. Route change
operation is relatively fast but making this change make it robust in
case the operation takes time.
Also changed image color and background color highlight color for
touch highlight.
Bug: webrtc:11778
Change-Id: I037b830f7a02b49bf292b8838bd245db585dbd22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179041
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31910}
This is a reland of 11dc6571cb4ff3e71dee1557dfff8d9076e108d3
One fix that makes Web Platform Tests pass in debug mode is applied.
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
This flag was used to protect an AdaptationListener from adapting again,
but since AdaptationListener was removed this guard is no longer needed.
Bug: webrtc:11834
Change-Id: If79e960f57aaf56ce2849329ff6737414fc60e9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181068
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31905}
Since the degradation preference can be fetch from the thread safe
provider, this removed the need to have 2 queues.
Bug: None
Change-Id: I94cb8b7d3d2950acfe0ad0a6d46edc038ed8e286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177523
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31898}
This reverts commit 11dc6571cb4ff3e71dee1557dfff8d9076e108d3.
Reason for revert: Breaks Chromium WPT tests
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
TBR=sakal@webrtc.org,kthelgason@webrtc.org,hta@webrtc.org,guidou@webrtc.org,marinaciocea@webrtc.org
Change-Id: Ibdc24f7d41e481293ca74ba6d1572de64f7e4654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:980879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181262
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31897}
It was not used by any class and all future uses can use the
VideoSourceRestrictionsListener.
Bug: webrtc:11834
Change-Id: I5c71b93cc503f458dce0ccdd78b91b5a1debc56d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181062
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31896}
This way can double adapt right away instead of relying
on the qp scaler checking soon into the future.
Bug: webrtc:11830
Change-Id: I8e878168303cf6a4c3edcf3997dd8ac2413a4479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181060
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31895}
This adds RtpTransceiver.StopStandard(), which behaves according to
the specification at
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
It modifies RTCPeerConnection.getTransceivers() to return only
transceivers that have not been stopped.
Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
Bug: chromium:980879
Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31893}
This is a followup to https://webrtc-review.googlesource.com/c/src/+/180883
and sets the correct visibility so that the enum is usable
for e.g classes overriding NetworkChangeDetector.
Bug: webrtc:11825
Change-Id: I2436a9634e0f7d45af195373e4f0a7ec8749c9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181072
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31883}
This patch adds an interface for os/firmware to set a network
preference NOT_PREFERRED / NEUTRAL that can be picked up by
an IceController and used when selection ice candidate pair.
The patch exposes this using an Android Intent based interface.
BUG: webrtc:11825
Change-Id: Ic12b6bf704fde7f9c912020dd7bc79ccae4613ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180883
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31877}
Added error checking in AudioIngress and AudioEgress to detect situations where codecs have not been set; added additional unit tests for VoipCore
Bug: webrtc:11251
Change-Id: Ibd57e518892c76e2865b844ba866e380a565dd6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180229
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31874}
This CL adds the possibility to log metainformation about
decoded frames in RTC event log, including encoding parsing
and tests. It will be wired up in a followup CL.
Bug: webrtc:8802
Change-Id: Ied598b266513d0f63fce0484d741af1782607e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181061
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31873}
The == operator needs to take the red_payload_type into account as
well.
BUG=webrtc:11640
Change-Id: Iae285643f60ca92704d17c0c088d8fb4b3cdc180
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178815
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31869}
this makes the association between opus and red a bit more obvious.
Also it allows access to the opus payload type which might be
used in the fmtp line in a future CL
BUG=webrtc:11640
Change-Id: I04e0648aedf049d103e3c3481c8712dfc9b79538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31868}
stats() method on EmulatedEndpoint has to be called from network
emulation internal task queue and user has no access to that task queue,
so user can't call this method. Because of that remove it from public
API and keep it only on implementation.
Bug: webrtc:11756
Change-Id: I2fb7256abe94d6900965512f90c6a53a0708a7b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31867}
