This reverts commit f342d6054ad984b7b80df2afe349c3bbb5f1d5b8.
or "Reland "Use non-proxied source object in VideoTrack.""
Reason for revert: Didn't resolve the downstream issues.
Original change's description:
> Revert "Reland "Use non-proxied source object in VideoTrack.""
>
> This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.
>
> Reason for revert: This is a partial revert as we're tracking down
> the source of the downstream issues. This CL reverts the use of
> `internal()` for methods that relate to the source sink.
>
> Original change's description:
> > Reland "Use non-proxied source object in VideoTrack."
> >
> > This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
> >
> > This reland doesn't contain the AudioTrack changes (see original
> > description) that got triggered in some cases and needs to be
> > addressed separately.
> >
> > Another change in this re-land is that instead of the `state` property
> > of the VideoTrack be marshalled to the signaling thread, it's readable
> > from the calling thread. Previously this was marshalled to the worker
> > and the original changed that to the signaling thread (same as for
> > AudioTrack) - but in case that's causing downstream problems this reland
> > uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> > VideoTrack proxy.
> >
> > Original change's description:
> > > Use non-proxied source object in VideoTrack.
> > >
> > > Use the internal representation of the video source object from the
> > > track. Before there were implicit thread hops due to use of the proxy.
> > >
> > > Also, override AudioTrack's enabled methods to enforce thread
> > > expectations.
> > >
> > > Bug: webrtc:13540
> > > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35911}
> >
> > Bug: webrtc:13540
> > Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35979}
>
> TBR=tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I4d8e3aced019215b97a6263cafa2a7488cd118be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:13540
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251661
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35983}
# Not skipping CQ checks because original CL landed > 1 day ago.
Using "no-try" since a follow-up revert is also needed to get the bots
to turn green.
No-try: true
Bug: webrtc:13540
Change-Id: I361fca6949c01200d9d706749e7e825eb5b4fc1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251685
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35991}
Adds new class DecodeSynchronizer that will coalesce the decoding
of received streams on the metronome. This feature is experimental and
is backed by a field trial WebRTC-FrameBuffer3.
This experiment now has 3 arms to it,
"WebRTC-FrameBuffer3/arm:FrameBuffer2/": Default, uses old frame buffer.
"WebRTC-FrameBuffer3/arm:FrameBuffer3/": Uses new frame buffer.
"WebRTC-FrameBuffer3/arm:SyncDecoding/": Uses new frame buffer with
frame scheduled on the metronome.
The SyncDecoding arm will not work until it is wired up in the follow-up
CL.
This change also makes the following modifications,
* Adds FakeMetronome utilities for tests using a metronome.
* Makes FrameDecodeScheduler an interface. The default implementation is
TaskQueueFrameDecodeScheduler.
* FrameDecodeScheduler now has a Stop() method, which must be called
before destruction.
TBR=philipel@webrtc.org
Change-Id: I58a306bb883604b0be3eb2a04b3d07dbdf185c71
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250665
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35988}
This is because it's the default flag used in the recipes to dump a json output.
This CL also fixes some python3 lint issues in mb.py.
Bug: webrtc:13594
Change-Id: I9275b5da0963f801d3191703c2eb72d90befb5d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248142
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35986}
The scalability mode could be something invalid set by user, in this
case, |num_spatial_layers| should not be updated.
Bug: chromium:1292923
Change-Id: I78e1a6f12cf6d165597205608e4c124117a3d01b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#35985}
This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.
Reason for revert: This is a partial revert as we're tracking down
the source of the downstream issues. This CL reverts the use of
`internal()` for methods that relate to the source sink.
Original change's description:
> Reland "Use non-proxied source object in VideoTrack."
>
> This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
>
> This reland doesn't contain the AudioTrack changes (see original
> description) that got triggered in some cases and needs to be
> addressed separately.
>
> Another change in this re-land is that instead of the `state` property
> of the VideoTrack be marshalled to the signaling thread, it's readable
> from the calling thread. Previously this was marshalled to the worker
> and the original changed that to the signaling thread (same as for
> AudioTrack) - but in case that's causing downstream problems this reland
> uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> VideoTrack proxy.
>
> Original change's description:
> > Use non-proxied source object in VideoTrack.
> >
> > Use the internal representation of the video source object from the
> > track. Before there were implicit thread hops due to use of the proxy.
> >
> > Also, override AudioTrack's enabled methods to enforce thread
> > expectations.
> >
> > Bug: webrtc:13540
> > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35911}
>
> Bug: webrtc:13540
> Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35979}
TBR=tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I4d8e3aced019215b97a6263cafa2a7488cd118be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13540
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251661
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35983}
Note that this will work on all platforms but is critical for windows due to the use backslash in the filesystem.
Bug: webrtc:13607
Change-Id: Ie9a9987f1382133792c85820d38b770fadc0fff5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251442
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35982}
Also expands integration_test_helpers to deal with multiple
datachannels.
The bug has not yet been triggered.
Bug: webrtc:13668
Change-Id: I82a0fdae0cc32815c250a691b56c614bfd6d606b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251602
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35980}
This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
This reland doesn't contain the AudioTrack changes (see original
description) that got triggered in some cases and needs to be
addressed separately.
Another change in this re-land is that instead of the `state` property
of the VideoTrack be marshalled to the signaling thread, it's readable
from the calling thread. Previously this was marshalled to the worker
and the original changed that to the signaling thread (same as for
AudioTrack) - but in case that's causing downstream problems this reland
uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
VideoTrack proxy.
Original change's description:
> Use non-proxied source object in VideoTrack.
>
> Use the internal representation of the video source object from the
> track. Before there were implicit thread hops due to use of the proxy.
>
> Also, override AudioTrack's enabled methods to enforce thread
> expectations.
>
> Bug: webrtc:13540
> Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35911}
Bug: webrtc:13540
Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35979}
Document with a comment the suspected place that could cause a bug.
Also fix an error in previous role observation code.
Bug: webrtc:13668
Change-Id: Id7f6af6905d90f7974b5570145c201c8339aaf72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251388
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35973}
This is important for writing tests that affect the DTLS role.
Bug: webrtc:13668
Change-Id: I5d9a93eca7996a8b74cdcfe412f59a85892e1ec1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251389
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35971}
Moving the template specialization into the header causes ODR
violation when the header file is included in other units. Making
the specialization inline to avoid this problem.
Bug: chromium:1291247
Change-Id: I090548c1c3dd07a8c46b87ae90ebdd45a60a5cde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251200
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35969}
This is a reland of 3ed36c0521546881656c73984456485dcab16205
Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}
Bug: webrtc:13540
Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35967}
Also deleted iwyu script that was not maintained, and deleted
some options that made the script more complex.
Bug: none
Change-Id: I39d8eaa37f12c72ddc127ae145e6a3a80f328316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251384
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35966}
A pointer to the transport controller is now maintained on
both the network thread and the signaling thread. We use
thread specific accessors to make it explicit which copy we
are accessing at any given time.
We also move the initial offerer value to the SDP offer/answer
class; this is determined on the basis of SDP offer/answer, so
there is no need to hop to the network thread for that.
Work in progress.
Bug: webrtc:9987
Change-Id: Idbe5a7fbf44f667adcd119e486133cf6e43ab1f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251382
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35965}
This reverts commit 3ed36c0521546881656c73984456485dcab16205.
Reason for revert: Breaks downstream project.
Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}
TBR=ilnik@webrtc.org,tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Ieb7235d88c808c78ad0847403be991d4dce1ace6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13540
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251383
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35963}
This is a reland of 325789c4576b60147ee1ef225d438cbb740f65ff
Original change's description:
> Mark all bool conversion operators as explicit
>
> An explicit bool conversion operator will still be used implicitly
> when an expression appears in "bool context", e.g., as the condition
> in an if statement, or as argument to logical operators. The
> `explicit` annotation prevents conversion in other contexts, e.g.,
> converting both a and b to bool in an expression like `a == b`.
>
> Bug: None
> Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35927}
Bug: None
Change-Id: Ie057dfc8c0b5c498e2c8daff7620172c89f0e011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35962}
The peerconnection target now has no files, which means that no
target that needs .h files depends on it. This is good.
Bug: webrtc:13634
Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35961}
This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
The difference from the original CL is that a check for
`state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
This caused a side effect that registering the sink while the source
was in an "initializing" state, failed. The last remaining state
however, is `kEnded` - but since there's no logic in the class around
the expected value of the states, the check inside of AddSink()
doesn't provide an additional value - it's rather a surprise for
developers if it doesn't succeed. So, now removed.
Original change's description:
> Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
>
> This simplifies the logic in these classes a bit, which makes upcoming
> change easier. The `stopped_` flag in these classes was essentially
> the same thing as `media_channel_ == nullptr`, which is what's
> consistently used now for the same checks.
>
> Bug: webrtc:13540
> Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35907}
Bug: webrtc:13540
Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35958}
In case we fail to import a DMA-BUF with given modifier, we can try to
drop the modifier we failed to use and renegotiate stream parameters
in order to use a different modifier or fallback to shared memory buffers.
Bug: chromium:1290566
Change-Id: I617513bdd67a43f62b647a172e0c166af138b3f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249798
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35957}
This allows us to keep always some frame around so we can return it
everytime consumer asks us to capture a frame as before we either
returned current frame or nothing as there was no new frame available.
This will be needed in order to support mouse cursor separately as
DesktopAndCursorComposer requires frame everytime, even if it's the
same one as before so we can combine it with the mouse cursor.
Bug: webrtc:13429
Change-Id: Ice87968846870c0a880ab469d9e052b4978e658c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239362
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35956}
The fake network configs are now specified using just two field trials:
WebRTC-FakeNetworkSendConfig and WebRTC-FakeNetworkReceiveConfig.
Both of them have the following parameters from
BuiltInNetworkBehaviorConfig:
* queue_length_packets // Queue length in number of packets.
* queue_delay_ms // Delay in addition to capacity induced delay.
* delay_standard_deviation_ms // Standard deviation of the extra delay.
* link_capacity_kbps // Link capacity in kbps.
* loss_percent // Random packet loss.
* allow_reordering // If packets are allowed to be reordered.
* avg_burst_loss_length // The average length of a burst of lost packets.
* packet_overhead // Additional bytes to add to packet size.
* codel_active_queue_management // Enable CoDel active queue management.
Plus:
* duration // For how long to use this config before progressing.
Example:
WebRTC-FakeNetworkSendConfig/queue_delay_ms:66|1,loss_percent:1|0,link_capacity_kbps:200|10000,queue_length_packets:10|100,duration:15s|45s/
This creates two configs:
1. For 15s, apply 66ms delay, 1% loss, 200kbps bandwidth, 10 packet queue size
2. For 45s, apply 1ms delay, 0% loss, 10Mbps bandwidth, 100 packets queue size
(then repeat)
Bug: webrtc:13655
Change-Id: I0524f572de480731df4d414724203772182c628b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251043
Reviewed-by: Stefan Holmer <holmer@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35952}
This enum is no longer needed. Also moving the last piece of code from
common.h to audio_processing_impl.h, allowing to delete common.h.
Bug: chromium:1271981, b/217349489
Change-Id: If115336c36d6d7b5845a903e421c18aebfe434ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251242
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35946}
This collision can occur when we have
asymetrical send and receive codecs. This is the case in the current
code base with the VP9 codec familly but is not visible untill more
codecs are added.
Added Nutanix Inc. to AUTHORS.
Bug: chromium:1291956
Change-Id: I09d3f76161d984d2a3edf721639753bffd4947b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250034
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35944}
