The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Part 1 of the migration towards merging the types.
Any method that could belong to the Codec type was moved, the others
are deprecated.
Alternatives to the AudioCodec and VideoCodec constructors are introduced
to allow creating objects of an indefinite type without having to
reference the old classes.
Bug: webrtc:15214
Change-Id: I20e1aa32962821cad98e9a92c2ec86f8f75e5dd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40213}
This completes the split-channel work for the Video side.
Note: For ease of review, the implementations in the .cc
file have not been sorted between sender and receiver. This
can be done in a later purely-editorial CL.
Bug: webrtc:13931
Change-Id: I36cf015d5facb1eed368070cb204a8763ac19a9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307180
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40207}
The intent is that this object can be used instead of VideoMediaChannel,
clearing the way for decomposing VideoMediaChannel into send and
receive classes.
This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines.
Bug: webrtc:13931
Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40173}
There are no common functions between MediaSendChannelInterface
and MediaReceiveChannelInterface except media_type().
This allows us to remove the common superclass for the two interfaces,
making for a simpler class structure.
Bug: webrtc:13931
Change-Id: I82a12ca31f0dc62d7bd97bdda34ca37e59a5fd55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40154}
This change allows us to remove one static_cast from tests that
was problematic for another refactoring.
Bug: webrtc:13931
Change-Id: I8e1b5cecadd806b266b6c115b56b18b9613cbe82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306500
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40144}
Prior to this CL, the video `jitterBufferDelay` stat was the accumulated current delay, which is a smoothened version of the target delay. This is not correct according to the spec [1]. Rather, the stat should be the accumulated time spent in the jitter buffer, for all emitted frames. This CL fixes this spec compliance problem.
Expect changes to test metrics and product monitoring as this CL rolls out.
[1]: https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
Tested:
1. Go to https://jsfiddle.net/jib1/0L6duga2/show
2. Apply 2.0 seconds of video delay.
3. Notice that "Video jitter buffer delay" is slightly less than 1990ms. (2000ms playoutdelayhint - 10ms render delay - Xms decode delay).
Bug: webrtc:15085
Change-Id: I42805faafd7dd3bcdcf3ad08e751e08d6de38906
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304521
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40138}
by not starting the receive stream whenever it is creating.
Instead, this is controlled by the direction of the media content.
BUG=webrtc:11013
Change-Id: Iaaa0ac0aa9f90a4be776a1348f53a0f9c2b84d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40064}
The field trial has been enabled-by-default for several years, I
suspect it was needed during its development but there doesn't seem to
be any reason to maintain it going forward.
Its very existence blocks our long term objective to have our APIs
behave according to the W3C standards and any apps still depending on
it, if there are any, should make sure to use the APIs correctly
instead. I assume they already do any any references to this is us
forgetting to clean things up.
Bug: webrtc:15161
Change-Id: I4a6a44a15219d2e045f3d8d857b5197a064f049c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304660
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40025}
This reduces dependency on the struct RTCPReportBlock and would allow to
delete it in favor of class ReportBlockData
Bug: None
Change-Id: Ia46a2516e26453724eed2e499f475f65df6cd3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304163
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39990}
* retransmittedBytesReceived
* retransmittedPacketsReceived
added to the specification in
https://github.com/w3c/webrtc-stats/pull/735
BUG=webrtc:15096
Change-Id: I6770e5d8d09ac1c2693c918fd943b0ab257ec7ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295260
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@{#39959}
This removes all "_WAIT" operations in the tests and all uses
of kTimeout for loop+poll check for values.
On my linux box, this also reduces the time it takes to run all
./rtc_media_unittests (in parallel) from about 3500ms, to ~50ms.
(longest running test was WebRtcVideoChannelBaseTest.GetStats)
Bug: none
Change-Id: If544aa10cb0281cb5e5e51fb654db5f45de871da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302343
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39918}
This reverts commit 75990b9a8f98ea2d597a31472fb778ec4d55f698.
Reason for revert: Breaks downstream, a use case of having three VP9
encodings, scalability mode only specified on the first layer
(L2T2_KEY) and the other two layers not having a scalability mode but
also being active=false appears to trigger a DCHECK in
call/rtp_video_sender.cc:501. More investigation needed
Original change's description:
> Ship ability to opt-in to VP9/AV1 simulcast.
>
> With this unflagging, an app can opt-in to simulcast when using multiple
> encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This
> ensures backwards-compat with apps relying on 3 encodings to mean SVC
> who traditionally have not specified scalabilityMode.
>
> It fixes the spec/API bug of asking for simulcast and not getting
> simulcast. The field trial exists only as a kill-switch with a TODO to
> remove it.
>
> This ships initial support, however note that the VP9/AV1 simulcast uses
> SimulcastRateAllocator (just like VP8/H264 simulcast). This rate
> allocator uses more kbps than SvcRateAllocator. This should be revisited
> to avoid significant higher bitrates, for example when comparing VP9
> simulcast to VP9 SVC.
>
> Shipping the ability for apps to opt-in makes it easier to exercise
> these new code paths and allows initial feedback from developers, but
> due to the high bitrate (= same bitrate as VP8/H264 simulcast today)
> many apps may find that VP9 SVC is still more beneficial for BW reasons.
>
> Bug: webrtc:14884, webrtc:15005
> Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39601}
Bug: webrtc:14884, webrtc:15005
Change-Id: Ic8f77e6a2971f493d6cd8c23faecd435058a8847
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298440
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39605}
With this unflagging, an app can opt-in to simulcast when using multiple
encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This
ensures backwards-compat with apps relying on 3 encodings to mean SVC
who traditionally have not specified scalabilityMode.
It fixes the spec/API bug of asking for simulcast and not getting
simulcast. The field trial exists only as a kill-switch with a TODO to
remove it.
This ships initial support, however note that the VP9/AV1 simulcast uses
SimulcastRateAllocator (just like VP8/H264 simulcast). This rate
allocator uses more kbps than SvcRateAllocator. This should be revisited
to avoid significant higher bitrates, for example when comparing VP9
simulcast to VP9 SVC.
Shipping the ability for apps to opt-in makes it easier to exercise
these new code paths and allows initial feedback from developers, but
due to the high bitrate (= same bitrate as VP8/H264 simulcast today)
many apps may find that VP9 SVC is still more beneficial for BW reasons.
Bug: webrtc:14884, webrtc:15005
Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39601}
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread. Avoid this
by posting only if invoked on a diffferent sequence.
TESTED=presubmit + local Meet calls.
Bug: webrtc:137439
Change-Id: Ic5dd99e5fbb843ad4c54d4466138135ae81596cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295867
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39471}
This allows MediaChannel to know whether it's being used
for sending, receiving, or both. This is a preparatory CL
for landing the split of MediaChannel usage into sending and
receiving objects.
Bug: webrtc:13931
Change-Id: If518c8b53d5256771200a42e1b5f2b3321d26d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292860
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39283}
This is used when an unsignaled stream with a known payload type is received and later a RTX packet is received.
Bug: webrtc:14817
Change-Id: I29f43281cec17553e1ec2483e21b8847714d2931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291328
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39243}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
This ensure transport feedback is sent for RTX packets that are received before media payload packets.
Bug: webrtc:14795, webrtc:14817
Change-Id: I6a2579b87c8863e003decb2b2559ef51a852cadb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291119
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39174}
Fallback to a default value if the scalability mode is unset or not supported by the codec.
The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set).
Bug: webrtc:11607
Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39074}
With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused.
Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid.
Bug: webrtc:7135, webrtc:14795
Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39058}
This is in preparation for splitting MediaChannel into sender and
receiver channels, with independent objects.
Bug: webrtc:13931
Change-Id: I8e34b0c80b4d76132394efcda658a8face3ab873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288750
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38998}
This is a reland of commit 97ba853295578975a04fc504315cccd465f9f0bd
This cl did not cause the regression in Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks. Real culprit reverted in https://webrtc-review.googlesource.com/c/src/+/290502.
Original change's description:
> Remove use of ReceiveStreamRtpConfig:transport_cc
>
> With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
> I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
>
>
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
>
> Bug: webrtc:14802
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38980}
Bug: webrtc:14802
Change-Id: Ib98e61413161d462da60144942cdb0140e12bc42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38997}
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Bug: webrtc:14802
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38980}
This CL removes a couple more opportunities for client code
to interact directly with the MediaChannel implementation classes.
No-try because of infra failure.
No-Try: true
Bug: webrtc:13931
Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38935}
The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.
Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
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}
The code for determining outbound-rtp.active assumed, as the spec says,
that we have one RtpEncodingParameters per RTP stream. Unfortunately
SVC is currently implemented as one RtpEncodingParameters per SVC
layer. This causes a discrepency where we do correctly only have one
outbound-rtp stats object, but the lookup to check whether or not we are
"active" needs to look at more than a single encoding.
The bug is that if SVC layers are {inactive, active, active} then
stats reports outbound-rtp.active: false. With this fix, active: true is
reported if ANY of the SVC layers are active.
For singlecast or simulcast this CL has no change in behavior. In these
cases we have the same number of outbound-rtp and encodings and a simple
ssrc lookup does work.
The fix is exercised by unit tests and has also manually been confirmed:
- Singlecast tested by https://jsfiddle.net/henbos/nvd6p4j1/.
- Simulcast tested by https://crbug.com/webrtc/14628#c11.
- SVC tested by Google Meet and chrome://webrtc-internals/.
Bug: webrtc:14628
Change-Id: Ib89945caf29c8f4b85dd8a1120dcf8279296e4a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282222
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38569}
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}
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).
The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.
Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
The experiment has been approved for a full launch. Changing the
default value so that no decoder is created before the stream starts.
All decoders are created lazily on demand when we receive payload
data of the corresponding type.
Bug: chromium:1319864
Change-Id: Ifb412bbe49a7577a45c340496d5b8572ebc1ba44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277120
Auto-Submit: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38232}
This cl/ is a NOP refactoring,
moving the EncoderStreamFactory from within webrtc_video_engine.cc
into own file in video/. simulcast.cc is collateral.
Bug: webrtc:14451
Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38224}
This way we're sure instantiation, configuration and decode calls all
happen on the decoder queue - making thread checking easier in the
actual decoder classes.
Bug: None
Change-Id: Ia98f47009f26b34eb8dad2ee0b4ddcde082d1994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272022
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37825}
...for payload type changes and avoid recreating the main video stream.
Bug: webrtc:11993
Change-Id: I03e44ff25d88caeb082a2e44b2e802d3b9392feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269244
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37666}
In WebRtcVideoChannelBaseTest.EncoderSelectorSwitchCodec a mock encoder
selecter or stack allocated and then registered with the channel.
Since this test uses real-time clocks/threads, there is a chance that
the selector callback will be called after the mock goes out of scope,
but before the test had time to be torn down.
This CL fixes that by simply de-registering the callback before the
end of the test.
Bug: b/239855550
Change-Id: Ibb38a914933494fd04c963b9a13f2cc4aee160d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269402
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37618}
This is a reland of commit 3afb8e24311dc1297150d4011894b6cb00841735
Patchset 1 is the original CL. Patchset 2 contains a fix:
Depending on call site, the number of spatial layers for VP9 might be
signalled in three different ways. One of them was afaict only used in
out perf tests, and resulted in the max bitrate being incorrectly
capped.
The fix now checks that field too.
Original change's description:
> When VP9 SVC is used, use SvcConfig to set max bitrate for the stream.
>
> Currently, a default max bitrate is determined within WebRtcVideoEngine,
> which maxes out at 2.5Mbps - and that limits the max bitrate deteremined
> by SvcConfig for resolutions above 720p.
>
> This does not affect simulcast, as WebRtcVideoEngine already knows to
> trust the rate allocation in simulcast.cc instead.
>
> Bug: webrtc:14017
> Change-Id: I0c310a6fd496e9e5a10eae45838900068aa1ae2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267160
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37370}
Bug: webrtc:14017, webrtc:14234
Change-Id: Idcaf4321a20c917e4049522c577336ddcfc7ffbb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267860
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37446}
