This reverts commit d926cf63b57128d9ea9a8d1054f853b4fe82e6dd.
Reason for revert: Breaks simulcast testing in Canary, to be relanded once the chrome part of the fix is landed as well.
Original change's description:
> [InsertableStreams] Fix simulcast: set frame transformer for all streams
>
> The transformer was previously moved into the config of the first stream
> which resulted in incorrect behavior for simulcast. Use the transformer
> in all the streams.
>
> Bug: chromium:1065838
> Change-Id: Iea340443da8cd4de32953bb24d3e6a07a275ae2a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173026
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31018}
TBR=mflodman@webrtc.org,marinaciocea@webrtc.org
Change-Id: Ib0f869ae617329eb2532b613741b6050bd3ba2a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1065838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173181
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31033}
Our implementation accepts TCP candidates with a missing tcptype
field, treating this as a passive candidate.
However, if you try to convert such a candidate to SDP and back,
which chromium started to do at some point, this was resulting in an
error. This CL fixes that.
Bug: webrtc:11423
Change-Id: Iec48d340f421f63f2b7a16c9496ea92ccd165981
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172020
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31026}
The RtpVideoStreamReceiverFrameTransformerDelegate::IsKeyFrame()
implementation was relying on the EncodedFrame::is_keyframe() API, which
checks the number of references to a frame. However the number of
references is not updated until after the frame is managed by the
receiver, after the frame has been transformed.
Update the delegate's IsKeyFrame() implementation to use the type of the
frame instead. The frame type is updated before transforming the frame,
on parsing the generic descriptor.
Bug: chromium:1068468
Change-Id: I84dadaecb1cd485262c2f1681dfa653d84693f69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173025
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31024}
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.
Reason for revert: Speculative revert: breaks downstream project
Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
>
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
>
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
>
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
>
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
>
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
>
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}
TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
The transformer was previously moved into the config of the first stream
which resulted in incorrect behavior for simulcast. Use the transformer
in all the streams.
Bug: chromium:1065838
Change-Id: Iea340443da8cd4de32953bb24d3e6a07a275ae2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173026
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31018}
This CL removes the AudioFrame-based APIs from the AudioProcessing
interface.
Bug: webrtc:5298
Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31016}
The TaskQueuePacedSender today has some inefficiencies:
* Enqueuing a packet will trigger a MaybeProcessPackets() call, but it
won't actually run immediately even if it should - instead it will
schedule a new call in at least 1ms. This incurs delays and extra
CPU overhead.
* Sometimes thread wakeups are scheduled simply in order to do
book-keeping: ProcessPackets() will be called when the media debt has
gone down to 0 even if there is no packet in the queue, in order to
check if we should send padding.
This CL fixes that by called ProcessPackets() immediately if it is
actually time to do so, and by immediately determining when padding
should be sent without having a separate call to drain media debt.
Bug: webrtc:10809
Change-Id: I4870e86e6de2ce4197463fd5b788ad4717fc7177
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172842
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31010}
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
This CL adds several field-trial-based overrides for parameters related
to AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11487,chromium:1067597
Change-Id: Ica9613db74240687fc85efe059874ef8c20aa7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172844
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31007}
Non-paced audio should be sent "immediately", but in several places that
was determined by looking at the current time - which can lead to
inconsistencies.
E.g. if a packet is enqueued and ProcessPackets() is called 1ms later,
the pacer should see NextSendTime() as 1ms ago, so that buffer levels
are cleared at the right pace.
Bug: webrtc:10809
Change-Id: I04a169f3df3e28a5c8ef7fa8a042b9c482c307ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172845
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31002}
Especially with the TaskQueuePacerSender, it is very common that a
single packet is added to the queue and then immediately removed as
it gets sent to the network.
This CL adds a fast-path for that case, that avoid creating book-
keeping in the form of stream-priorities and timestamp sets etc.
Functionally, it should be a noop, but hopefully it can save a few
CPU cycles.
Bug: webrtc:10809
Change-Id: Idaa06b4f8d1da444fce78cc742e2ab52f9efe815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172090
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31001}
This CL adds two kill-switches to the AEC3 code to be used as
safe fallbacks to increase AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11487,chromium:1067597
Change-Id: I7f9f78db4964990bcdfa9adae6ef36a56bce7224
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172840
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30998}
This patch is a follow up to https://webrtc-review.googlesource.com/c/src/+/172582
and change so that a switch from CELLULAR_X to CELLULAR_Y does not
trigger OnNetworkChange.
This is needed as the OnNetworkChange signals triggers
BasicPortAllocator to rescan all networks and generate new candidates.
The actual adapter type change is still possible to react on using
SignalTypeChanged.
BUG: webrtc:11473
Change-Id: Icc1a945b8a4df1714c6ec4b02ec759ecada92d7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172802
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30992}
It's not expected this will make a difference, since the packet should
be read from the queue if possible as soon as it's added to it.
But we're doing this as an added precaution in case we overlooked
something. See linked bug.
Bug: chromium:1063834
Change-Id: I7a3a6d86a97683cbcbeed5ef1aaa8090cf6bf8c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172661
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30990}
