This moves the thread hop for network events, from BaseChannel and
into Call. The reason for this is to move the control over those hops
(including DeliverPacket[Async]) into the same class where the state
is held that is affected by those hops. Once that's done, we can start
moving the relevant network state over to the network thread and
eventually remove the hops.
I'm also adding several TODOs for tracking future steps and give
developers a heads up.
Bug: webrtc:11993
Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33138}
Functionality wise, there should be no change with this CL, aside
from updating tests to anticipate OnPacketReceived to handle the packet
asynchronously (as already was the case via BaseChannel).
This only removes the network->worker hop out of the BaseChannel
class into the WebRTC MediaChannel implementations. However, it updates
the interface contract between BaseChannel and MediaChannel to align
with how we want things to work down the line, i.e. avoid hopping to
the worker thread for every rtp packet.
The following steps will be to update the video and voice channel
classes to call Call::DeliverPacket on the network thread and only
handle unsignalled SSRCs on the worker (exception case).
Bug: webrtc:11993
Change-Id: If0540874444565dc93773aee89d862f3bfc9c502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33040}
Higher range of IDs for peer connection has been exhausted.
Adding AV1 to lower range as it was blocking enabling
libaom by default.
This is blocking crrev.com/c/2617229
Bug: chromium:1095763
Change-Id: If5135122954d00cc03afc563071aec99f145140b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201523
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32967}
It turned out that the negotiated rtp header extensions are not fully known in WebRtcVideoChannel::AddSendStream.
The cl also remove the unnecessary factory for creating VideoStreamEncoder.
Bug: webrtc:12000
Change-Id: If994c8deb69f3ce4212896d3ad757dac94c6e09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32916}
This is a reland of f08db1be94e760c201acdc3a121e67453960c970
Original change's description:
> Enable FlexFEC as a receiver video codec by default
>
> - Add Flex FEC format as default supported receive codec
> - Disallow advertising FlexFEC as video sender codec by default until implementation is complete
> - Toggle field trial "WebRTC-FlexFEC-03-Advertised"s behavior for receiver to use as kill-switch to prevent codec advertising
>
> Bug: webrtc:8151
> Change-Id: Iff367119263496fb335500e96641669654b45834
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191947
> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32639}
Bug: webrtc:8151
Change-Id: I36cbe833dc2131d72f1d7e8f96d058d0caa94ff9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195363
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32819}
Without this change, if the user disables QVGA and VGA streams via |active|
flags in SetParamters, the resulting stream would have too high min bitrate.
This would lead to bad performance and low quality adaptation rate.
Bug: none
Change-Id: I919a30bfb248c06747c989afe6965b3afaef2260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195325
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32706}
- Add Flex FEC format as default supported receive codec
- Disallow advertising FlexFEC as video sender codec by default until implementation is complete
- Toggle field trial "WebRTC-FlexFEC-03-Advertised"s behavior for receiver to use as kill-switch to prevent codec advertising
Bug: webrtc:8151
Change-Id: Iff367119263496fb335500e96641669654b45834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191947
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32639}
This CL does not aim at cleaning up simulcast/SVC configuration, just to make it possible to set the scalability mode for AV1. Implementing a codec agnostic SVC/simulcast API is a (big) project on its own.
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
BUG: webrtc:11607
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192541
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32631}
VideoBitrateAllocation is instead reported through the EncoderSink.
Enable VideoBitrateAllocation reporting from WebRtcVideoChannel::AddSendStream in preparation for
using the extension RtpVideoLayersAllocationExtension instead of RTCP XR.
Bug: webrtc:12000
Change-Id: I5ea8e4f237a1c4e84a89cbfd97ac4353d4c2984f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186940
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32347}
Currently, when only max bitrate available and min bitratea & target
bitrate are missing from encoding config, the target bitrate is decided
by the calculation from GetSimulcastConfig() according to width/height/qp.
The max bitrate doesn't play a role here other than ensure target < max.
This will make the target bitrate cap at some calculated number even
when control message gives much larger allocation through max bitrate.
In our cases, the L0 (at 180p) is capped at 80-90kbps even control
message gives L0's max bitrate over 300kbps. This under-use of bandwidth
happens to all layer other than top layer. Top layer will be compensated
with all the left bandwidth up to max at last.
Since in web api, we cannot pass down either min bitrate or target bitrate
(https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpEncodingParameters).
We propose a new logic to take max bitrate into consideration in this case,
use 3/4 max bitrate or calculated target bitrate whichever is larger.
Bug: None
Change-Id: I2234b4636daa379fd47d4bbe764cf8307b9a1ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186161
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32308}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
RequestEncoderFallback, RequestEncoderSwitch and
SetVideoCodecSwitchingEnabledRequest are now all called on the
worker thread. Before, the work already happened on that thread but
WebRtcVideoChannel adapted internally when needed.
With this CL, there are thread checks to make sure that these calls are
always made the same way, we don't need the async invoker and there
are fewer calls out from the encoder thread in VideoStreamEncoder
(reducing the chance of unintentional blocking).
Bug: webrtc:11908
Change-Id: If8738bc2a708a0fefc6fe850b32655f049f30bdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184603
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32151}
This turned out to be a bit complicated, mostly
related to the tests, but here's what's changed:
* No AsyncInvoker (and avoid ClearInternal) in
WebRtcVideoSendStream (WVSS)
* The reason it was there is due to a "design leak" from
VideoSourceSinkController/VideoStreamEncoder where the former uses
locks in all methods and is unaware of a threading model. That design
affected downstream objects, pushed the need for an async hop into
WVSS and added a lock.
A suggestion was made to address this in a follow-up change, here:
https://webrtc-review.googlesource.com/c/src/+/165684
* All methods in VideoSourceSinkController are now called on a known
and checked sequence and this CL removes the lock. This also makes
checking state consistent (i.e. calling a getter twice in a row on the
same sequence, will always return the same value, avoiding race with
other threads).
* Handling of reporting state changes from the encoder queue to the
VSSC, is done by VideoStreamEncoder.
* VideoSendStreamImpl is still instantiated on the incorrect thread [1]
but has two initialization steps [2]. The second one already runs on
the right thread. Addressing that TODO [1] is something we should do
but it has side effects to consider. For the purposes of this CL
the steps relating to the encoder (setting the sink pointer) have
been moved to [2].
[1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;l=94
[2] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;drc=f4a9991cce74a37d006438ec0e366313ed33162e;l=115
Bug: webrtc:11222, webrtc:11908
Change-Id: Ie46d46e3a52bbe225951b4bd580ecb8cc9cad873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184508
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32150}
This is a reland of 32ca95145c4636374266f5b5d4d1ac43658bc758
Fix includes not enabling the screenshare conference behavior on non
screenshare sources even if the flag is enabled.
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
Bug: webrtc:11310
Bug: chromium:1093819
Change-Id: Ic933f93a5c4bad20583354fe821f8a1170e911cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180802
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31847}
This reverts commit 32ca95145c4636374266f5b5d4d1ac43658bc758.
Reason for revert: Internal test failure
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
TBR=ilnik@webrtc.org,hta@webrtc.org,orphis@webrtc.org
Change-Id: I5ccb6e87594f491ba09fe6b837ee24d63db878ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11310
Bug: chromium:1093819
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180801
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31829}
Non-conference mode simulcast screenshares were mistakenly using the
conference mode semantics in the simulcast rate allocator, which broke
spec compliant usage in some situation.
This behavior should only be used when explicitly using the SDP entry
"a=x-google-flag:conference" in both offer and answer.
Bug: webrtc:11310, chromium:1093819
Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31828}
This fixes a crash that could happen if substreams exist but there is
no kMedia substream yet. There was an assumption that we either had no
substreams or at least one kMedia substream, but this was not true.
The correct thing to do is to ignore substream stats that are not
associated with any kMedia substream, because we only produce
"outbound-rtp" stats objects for existing kMedia substreams.
A test is added to make sure no stats are returned. Prior to the fix,
this test would crash.
Bug: chromium:1090712
Change-Id: Ib1f8494a162542ae56bdd2df7618775a3473419b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176446
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31442}
It seems possible that getStats() and merging RTX/FlexFEC substream
stats into media substream stats can race with the creation or
destruction of the media substream that the RTX/FlexFEC substream is
associated with.
In other words, the DCHECK that ensures that there exists a stats object
to merge into is not always valid. Because there is no media stats
object to merge in to, and outbound-rtp stats objects only exists per
media SSRCs, the sensible thing to do is to RTC_LOG and ignore the
substream stats.
Bug: webrtc:11545
Change-Id: I4061d7190da7ab8bd33fa1fd92c9d819f35d76c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31156}
Save the frame transformer set on unsignaled receivers, and set the
transformer when the ssrc becomes known.
Pass the receiver's ssrc on registering the transformed frame callback,
to associate separate frame transformer sinks for each receiver.
Bug: chromium:1065838
Bug: chromium:1065838
Change-Id: I2a214bdb6cb9a8012928a03f046f311c344370f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173201
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31051}
