DSCP is controlled by the spec-compliant API
RTCRtpEncodingParameters.networkPriority[1]. It already has a default
value that is the same as when DSCP is disabled.
- If you want non-default DSCP default values, you need to set
networkPriority and shouldn't need to set a non-standard googDscp flag
for it to have an effect.
- If you want the default DSCP value, you wouldn't change
networkPriority and so you don't care if enable_dscp is true... you'll
get the default regardless.
Drive-by: This CL also adds crbug references to other goog flags.
[1] https://w3c.github.io/webrtc-priority/#dom-rtcrtpencodingparameters-networkpriority
Bug: chromium:1315574
Change-Id: I15a0470fa04f55e2534cee0d240eeb03446c2de6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36550}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
This is a simple check for upcoming changes for media channels to
be able to check if the state on the network thread is consistent.
Bug: webrtc:11992
Change-Id: I8ed2d091ecf3869a66970fc4733aebf209c4ef82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35706}
Because of this (seemingly simple) change, I had to change the return
type of transport_name from `const std::string&` to `absl::string_view`
to handle the case when there's no transport assigned.
That in turn caused an avalanche of required updates.
Bug: webrtc:12230, webrtc:11993
Change-Id: I16ec6c6a5fc2f5f7c7de572355a3c6ca924bb9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244084
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35617}
This makes things slightly simpler for the time being as surrounding
code is being refactored. This also removes a PostTask which has the
effect of shrinking the window between the Pending/Complete
notifications slightly since there's no additional async task
for the 'complete' step.
Bug: webrtc:11993
Change-Id: Ia86779b21c6f87301f37d763f89ace722e06e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244081
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35609}
Removes a few constructors where similar ones existed.
Removes MediaConfig dependency from MediaChannel and fixes an iwyu.
Bug: none
Change-Id: I9e34a1da0852c3fb21222161fad315e70598db3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242966
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35608}
This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb.
Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2.
Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34897}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34930}
This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=hta,hbos,minyue
Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
jitterBufferDelay and jitterBufferEmittedCount are defined
in RTCMediaStreamStats for both audio and video.
But for video, they were not populated in RTCInboundRtpStreamStats.
Bug: webrtc:12910
Change-Id: I135d473f055ecfb2c39b078ccf18c1bb9bc4f210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34398}
cricket::SendDataParams is replaced by webrtc::SendDataParams.
cricket::DataMessageType is replaced by webrtc::DataMessageType.
The sid member from cricket::SendDataParams is now passed as an argument
to functions that used one when necessary.
Bug: webrtc:7484
Change-Id: Ia4a89c9651fb54ab9a084a6098d49130b6319e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217761
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33966}
Pending messages on network thread for MediaChannel, will be dropped
when the MediaChannel object is deleted (without blocking).
Remove MessageHandler inheritance from Channel since Post-ing to the
network thread has been removed from there.
Copy/pasted code for SendRtp/SendRtcp in WebRtcVideoChannel and
WebRtcVoiceMediaChannel consolidated in MediaChannel.
Bug: webrtc:11993
Change-Id: I05320eb7f86b98adba50ca5eb8b76b78f4111263
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217720
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33955}
It doesn't make sense to use negative values or 0 to disable the
feature, so we use an optional int value.
Values bigger than 65535 are clamped down.
Bug: webrtc:12730
Change-Id: I6bd9cd92f7d0a70a78cf5a7c91dca52c28d08ba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33954}
This CL mostly adds plumbing to get awareness of the network thread
to the media channel classes. Currently this pointer is only used
to DCHECK that `SetInterface` for the `NetworkInterface` pointer, is
called on the network thread. Follow up changes will establish that
most of the methods are called on the network thread and the mutex
in the MediaChannel base class, can be removed.
Most of the changes in the CL are in channel_unittest.cc. They're mostly
around updating the tests to incorporate the network thread in ways
that reflect how the classes are used in production. Another change is
to use accessor methods for the media channel instances instead of
caching potentially dangling pointers.
Bug: webrtc:11993
Change-Id: I8e2ed1bc23724e238554dbce386789d69660f7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217682
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33951}
This field was only used in RTP Data Channels and isn't needed anymore.
Bug: webrtc:6625
Change-Id: Ieaa7ae03ca3e90eb4ddec4d384f5a76cef1600cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215687
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33791}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
BaseChannel adds and removes receive streams on the worker thread
(UpdateRemoteStreams_w) and then posts a task to the network thread to
update the demuxer criteria. Until this happens, OnRtpPacket() keeps
forwarding "recently removed" ssrc packets to the WebRtcVideoChannel.
Furthermore WebRtcVideoChannel::OnPacketReceived() posts task from the
network thread to the worker thread, so even if the demuxer criteria was
instantly updated we would still have an issue of in-flight packets for
old ssrcs arriving late on the worker thread inside WebRtcVideoChannel.
The wrong ssrc could also arrive when the demuxer goes from forwarding
all packets to a single m= section to forwarding to different m=
sections. In this case we get packets with an ssrc for a recently
created m= section and the ssrc was never intended for our channel.
This is a problem because when WebRtcVideoChannel sees an unknown ssrc
it treats it as an unsignalled stream, creating and destroying default
streams which can be very expensive and introduce large delays when lots
of packets are queued up.
This CL addresses the issue with callbacks for when a demuxer criteria
update is pending and when it has completed. During this window of time,
WebRtcVideoChannel will drop packets for unknown ssrcs.
This approach fixes the race without introducing any new locks and
packets belonging to ssrcs that were not removed continue to be
forwarded even if a demuxer criteria update is pending. This should make
a=inactive for 50p receive streams a glitch-free experience.
Bug: webrtc:12258, chromium:1069603
Change-Id: I30d85f53d84e7eddf7d21380fb608631863aad21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214964
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33757}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33404}
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
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}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
The degradation preference is now based on the content hint of the track
if it's unspecified.
Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30691}
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.
The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
Add a new API in RTPSenderInterface, to be called from the browser side
to insert a frame transformer between the Encoded and the Packetizer.
The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in RTPSenderVideo, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169128.
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30642}
Refactor voice engine and video engine to use default methods instead of
treating 0 as a special value.
Bug: webrtc:8694
Change-Id: I47c211c6e870cdec737d6b0d05df29a9b534a011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158600
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30010}
