This allows subclasses of MediaSendChannel and MediaReceiveChannel
to derive from MediaChannelUtil without promising to implement
the interfaces.
Bug: webrtc:13931
Change-Id: I998de7566b343032c83cd6e5419f49349f41035f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40185}
This allows us to decouple implementation classes from the
MediaChannel class.
Bug: webrtc:13931
Change-Id: I22f166cac17c344f943a0382048e8086a193affa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40179}
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}
as the killswitch is no longer required.
BUG=webrtc:12194
Change-Id: Icb825012c50a93ec4dae49be5732d9e4c0adb89d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306182
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40149}
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}
These tests verify the ability to override either the old or the
new function, and get the expected results.
Bug: webrtc:13931
Change-Id: Iebd0c929eda73dea75f32b96eb91a64e059a3cf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294880
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40120}
This makes the handling somewhat more uniform, and is the same
for both video and audio channels.
Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
Change default value of is_active to false,
this means that VideoRenderer or other VideoSinks
added with default rtc::VideoSinkWants() does not
block usage of RequestedResolution, e.g JNI_VideoTrack_AddSink.
This problem occurs when attaching a VideoRenderer directly to
the sending VideoTrack (which is a great solution!). But the
VideoRenderer is "passive" and should not block adaptations
from RequestedResolution.
Bug: webrtc:14451
Change-Id: I2ab02596245c7b82bf94fe86f8788f458c7ea286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305024
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40105}
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}
Before the channel split, the RTP modes were set by reading the
configuration of the send codec. After the split, this is done
via the SetReceiverFeedbackParams function.
This CL adds caching those parameters so that they are applied
to receive streams created after the SetReceiverFeedbackParams call.
Bug: webrtc:13931
Change-Id: I92eb651e5dd1ec68aca7f6a162e3521eb835a11d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305021
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40056}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
* 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}
Assuming 15Mbps video bitrate at 30fps, a single frame is 62500 bytes.
Add to that some fluctuations in encoder output rate and capture fps,
and frames can easily become larger than 64kb.
Given enough bandwidth and the bursty pacer, it will not be uncommon to
send the entire frame in one batch - and if the send buffer is at 64kb
then you will likely get packetloss already in the IPC packet socket,
even before the packet has reached the network card!
It's not entirely clear what the optimal size is, but given that the
receive buffer size was increased from 64kb to 256kb for high bandwidth
receive scenarios and had negligible negative effects I think it's
pretty safe to bump the send buffer to match.
There is a field trial available that can be used as circuit breaker
in case things turn south: WebRTC-SendBufferSizeBytes
Bug: webrtc:14780
Change-Id: I6c786d993181a882e6dce832ff56dc92d2a8a341
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290985
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39942}
This avoid overflow when handling large input sizes, e.g.2016x1512, or 2592x1944.
Bug: webrtc:15030
Change-Id: I97d5fa163ce0fac4c47f21826656819e652efafe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300240
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39774}
which describe the existing behavior that necessitated the revert
b396e2b159b060791954495d68278a55e8f72092
Also change the fake media engine audio clockrate to 8000 instead
of 0 and the fake media engine video payload type to something but
0 as this value seems to be treated specially by the video engine
and is a payload type reserved for PCMU.
BUG=chromium:1051821
Change-Id: Ib0a345d59baba50a565f01685d240e41584367e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299000
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39699}
This avoids a couple of layers of error code conversion, reduces
dependency on cricket error types and allows us to preserve error
information from dcsctp. Along the way remove SendDataResult.
Bug: none
Change-Id: I1ad18a8f0b2fb181745b19c49f36f270708720c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298305
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39619}
This struct only contains two member variables now and there isn't
much value added by having it.
Low-Coverage-Reason: No change in coverage, CL modifies uncovered RTC_LOG lines.
Bug: none
Change-Id: I924d450f4c8f8e49b1cfeabaebee9fd5235a90cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297360
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39563}
This reverts commit 18c869bc36b342cd4a79947067e52a93a04a7808.
Reason for revert: Added a field trial that allows landing the code without affecting performance in prod.
This CL also incorporates subsequent CLs that also had to be reverted.
Original change's description:
> Revert "Use two MediaChannels for 2 directions."
>
> This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.
>
> Reason for revert: Quality regression detected.
>
> Original change's description:
> > Use two MediaChannels for 2 directions.
> >
> > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
> >
> > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
> >
> > Bug: webrtc:13931
> > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39340}
>
> No-Try: true
> Bug: webrtc:13931
> Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39445}
Bug: webrtc:13931
Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39494}
This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.
Reason for revert: Quality regression detected.
Original change's description:
> Use two MediaChannels for 2 directions.
>
> This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
>
> The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
>
> Bug: webrtc:13931
> Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39340}
No-Try: true
Bug: webrtc:13931
Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39445}
This reverts commit a087f6f1c842f1d70ad207b44c48321ab60d2d95.
Reason for revert: Needed to roll back other CL
Original change's description:
> Add plumbing for video NACK to be coupled between channels.
>
> Bug: webrtc:13931, webrtc:14920
> Change-Id: I451869e295e099a1d08c0c80e481decd53149f1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294382
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39373}
Bug: webrtc:13931, webrtc:14920
Change-Id: I19e176e75630313da470542e7ff1e89b6d717fc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295664
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39432}
following the removal of ISAC from the code base.
BUG=webrtc:14450
Change-Id: I6faab5391bf0ef563c5dcce0bd5d8a653a87d9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294523
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39378}
Also update the tests that depend on FakeMediaEngine.
Bug: webrtc:13931
Change-Id: Ia608c4ce68a29e45174b68ba0103af31e9a7d3d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39345}
This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
Bug: webrtc:13931
Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39340}
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 done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
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 a reland of commit 81aab488781c1a736c9d85ff1532631be2989523
See diff between Patch Set 1 and latest Patch Set.
The original CL broke this WPT[1] because getStats() with the receiver
as the selector stopped working in the event of unsignalled SSRCs due
to the receiver not knowing what the SSRC was.
This fix is to query media_channel_ for the unsignalled SSRC in the
event that the receiver does not know the SSRC.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html
Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}
Bug: webrtc:14175, webrtc:14811
Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39010}
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 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}
This is a reland of commit d49d49ad89e67d1a3c63fbc638af445af5648875
Fixed seconds to milliseconds conversion in VideoAnalyzer.
Original change's description:
> Report total and squared inter frame delays measured in OnRenderedFrame
>
> After https://webrtc-review.googlesource.com/c/src/+/160042 we ended up with two sets of metrics representing total and total squared inter frame delays: old is measured in OnDecodedFrame and new in OnRenderedFrame. Reporting of old metrics was unshipped in https://webrtc-review.googlesource.com/c/src/+/278100. The metrics are used for calculation of harmonic frame rate and are desired to be measured as close as possible to rendering. This CL removes calculation of inter frame delay metrics from OnDecodedFrame and reports the metrics calculated in OnRenderedFrame to the stats.
>
> Bug: webrtc:11108, b/261512902
> Change-Id: Ia21b321aab3a1ac0b6136dc0df7d95f2f0fd24c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286842
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38909}
Bug: webrtc:11108, webrtc:14792, b/261512902
Change-Id: Ic5d0bc4622ee0cb46b6c225cdddccc217200e794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288641
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38929}
This reverts commit d49d49ad89e67d1a3c63fbc638af445af5648875.
Reason for revert:
# Check failed: total_freezes_duration_ms_double <= total_frames_duration_ms_double (196 vs. 0.044783)
https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Mac%20M1%20Arm64%2012
it also breaks the metric 'freeze_duration_ratio':
https://chromeperf.appspot.com/report?sid=6e919d271ff5885c3fa6363dd255b9793d5e79332a9f202b725c33cc7d3da31a
Original change's description:
> Report total and squared inter frame delays measured in OnRenderedFrame
>
> After https://webrtc-review.googlesource.com/c/src/+/160042 we ended up with two sets of metrics representing total and total squared inter frame delays: old is measured in OnDecodedFrame and new in OnRenderedFrame. Reporting of old metrics was unshipped in https://webrtc-review.googlesource.com/c/src/+/278100. The metrics are used for calculation of harmonic frame rate and are desired to be measured as close as possible to rendering. This CL removes calculation of inter frame delay metrics from OnDecodedFrame and reports the metrics calculated in OnRenderedFrame to the stats.
>
> Bug: webrtc:11108, b/261512902
> Change-Id: Ia21b321aab3a1ac0b6136dc0df7d95f2f0fd24c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286842
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38909}
Bug: webrtc:11108, b/261512902, webrtc:14789
Change-Id: Ie0da33c1071c48c50bff6608830c9e2a5a928fb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288402
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38922}
After https://webrtc-review.googlesource.com/c/src/+/160042 we ended up with two sets of metrics representing total and total squared inter frame delays: old is measured in OnDecodedFrame and new in OnRenderedFrame. Reporting of old metrics was unshipped in https://webrtc-review.googlesource.com/c/src/+/278100. The metrics are used for calculation of harmonic frame rate and are desired to be measured as close as possible to rendering. This CL removes calculation of inter frame delay metrics from OnDecodedFrame and reports the metrics calculated in OnRenderedFrame to the stats.
Bug: webrtc:11108, b/261512902
Change-Id: Ia21b321aab3a1ac0b6136dc0df7d95f2f0fd24c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286842
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38909}
This detaches the implementation (which is still merged)
from the objects used to interface to it.
Bug: webrtc:13931
Change-Id: I872ee10e4ed9fa432bfa231f723af1d3989d79d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38906}
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}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
This allows callers to differentiate on whether they need the
channel for sending or receiving purposes.
Note: This CL is incomplete, in that many places cast the pointers
to the concrete subclasses "VideoMediaChannel" and "AudioMediaChannel", which are not split into sending and receiving APIs.
The long term goal is to make two MediaChannel-like class APIs, with distinct implementations, and let the RtpSender and RtpReceiver manage those objects, rather than keeping them in the RtpTransceiver.
Bug: webrtc:13931
Change-Id: I8d56defe2287bd6552b71571cc6a5ec842927fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287040
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38844}
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}
and do the resolution of rids to layers. This has no effect yet
since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer.
BUG=chromium:1354101
Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38565}
