Removes all unused features, reducing the exposed interface surface.
This makes refactoring and maintenance simpler as we can change
TransportFeedbackAdapter without making corresponding changes
to RtpVideoSender.
Bug: webrtc:9883
Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29649}
This CL sets the RTP stats callback on construction, by adding a field
next to the other observers in RtpRtcp::Configuration.
We can then remove the RegisterCallback() methods and the unused
GetCallback() method.
Bug: webrtc:11036
Change-Id: I4eb86ea63b4b2ebeff60b311ddf3bed06b279ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157169
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29504}
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
Downstream test now fixed.
As a precaution, also avoid DCHECKS for non-zero SSRC.
First patch set is reland, second makes checks more lenient.
Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}
Bug: webrtc:10774
Change-Id: I540b49a31a31e98d87f02ae04083d5206e71c1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157100
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29491}
This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.
Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.
This prepares for reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29490}
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
Downstream fixed, relanding.
Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}
TBR=nisse@webrtc.org
Bug: webrtc:10774
Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29486}
Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
remove them, make the members const, and remove now unnecessary locking.
Bug: webrtc:10774
Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29475}
use frame_type from the RTPVideoHeader instead of as an extra parameter
merge payload data and payload size into single argument
pass RTPVideoHeader by value (relying on copy elision)
Bug: None
Change-Id: Ie7970af3b198b83b723d84c7a8b047219c4b38c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29445}
This CL moves the various parameters in the the RTPSenderVideo ctor into
a struct, and adds the red/ulpfec payload types to it.
Once the downstream usage of SetUlpfecConfig() is gone, we can make
those members const and avoid locking in SendVideo().
Bug: webrtc:10809
Change-Id: I9a96ab5b2a4eb2997ebf4a3a3e3cd2eb5715fd79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155365
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29384}
This CL removes some unnecessary locking, since we are already
serialized by the lock in VideoStreamEncoder. A simple RaceChecker is
used to verify this.
We also remove the usage of RegisterPayloadType() and replace it with
a parameter in SendVideo instead. This way we are prepared for removing
the payload type map and lock entirely. Some usage still exists
downstream and needs to be removed before cleaning this up.
Bug: webrtc:10809
Change-Id: Ie90163f15d11c8843f3beaf9a0df0dd2a1fd5ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154700
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29372}
This interface/config field is now unused, let's remove it.
Bug: webrtc:10633
Change-Id: I56ff3d47ba784d973de411ada52ec9485bad9864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150531
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28978}
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.
The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.
Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
In some situation, we disable simulcast in the encoder pipeline without
changing the sender's RIDs and it would crash.
This should only happen now when requesting simulcast with VP9 codec,
for which you currently get SVC instead.
Bug: webrtc:10660
Change-Id: I4f3b3d7760aded8f0769f8357c03ed8580ea46fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145336
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28669}
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.
The functionality is negotiated using SDP.
It's added with a field trial that allow disabling the functionality in
case there's any issues.
Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
Zero bitrate caused division by zero in DCHECK for max bitrate.
Added unit tests to ensure that setting zero bitrate does not crash.
> Original change's description:
> > Cleanup of video packet overhead calculation.
> >
> > This CL updates the video packet overhead calculation to make it more
> > clear. This prepares for future work on improving the accuracy of the
> > calculation.
> >
> > Bug: webrtc:9883
> > Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28040}
Bug: webrtc:10674
Change-Id: I156d1ee5546ede7e43ae1d9a298dcaba6071230f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140890
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28212}
This reverts commit 35d4e43f169e7cb237bce9501db29ea4b69820cd.
Reason for revert: Breaks downstream.
Original change's description:
> Reland "Cleanup of video packet overhead calculation."
>
> This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
>
> The calculation was rewritten using the new Frequency type to
> avoid the division by zero error introduced by the previous CL.
>
> Original change's description:
> > Cleanup of video packet overhead calculation.
> >
> > This CL updates the video packet overhead calculation to make it more
> > clear. This prepares for future work on improving the accuracy of the
> > calculation.
> >
> > Bug: webrtc:9883
> > Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28040}
>
> Bug: webrtc:10674
> Change-Id: Ib5cb6f05cfa7d097f89ac6fdcf198f2fc1b26b58
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138219
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28194}
TBR=nisse@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: Ib6c3c123590b815c4be12966cdae02f91c61ab34
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10674
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140889
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28195}
This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
The calculation was rewritten using the new Frequency type to
avoid the division by zero error introduced by the previous CL.
Original change's description:
> Cleanup of video packet overhead calculation.
>
> This CL updates the video packet overhead calculation to make it more
> clear. This prepares for future work on improving the accuracy of the
> calculation.
>
> Bug: webrtc:9883
> Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28040}
Bug: webrtc:10674
Change-Id: Ib5cb6f05cfa7d097f89ac6fdcf198f2fc1b26b58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138219
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28194}
Using this flag, an encoder may inform the RTP sender module that
the packet is not elligible for retransmission. Specifically, it
may not be retransmitted in response to a NACK message,
nor because of early loss detection (see CL #135881).
Bug: webrtc:10702
Change-Id: Ib6a9cc361cf10ea7214cf672e05940c27899a6be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140105
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28169}
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats
In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
video streams, but the code is not wired up to make use of these stats.
In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.
[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067
Bug: webrtc:10456
Change-Id: Icd20452cb4b4908203b28ae9d9f52c25693cf91d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138065
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28071}
This reverts commit 890bc3069cbababa19b40ec02684253d60e051b2.
Reason for revert: Div by zero.
Original change's description:
> Cleanup of video packet overhead calculation.
>
> This CL updates the video packet overhead calculation to make it more
> clear. This prepares for future work on improving the accuracy of the
> calculation.
>
> Bug: webrtc:9883
> Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28040}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: Icbdfc7b9252f8f9aa8e7e97b85b04171a27935e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138212
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28049}
The following private methods needlessly took a reference to the
RtpConfig on which they had worked, which was itself a member.
* ConfigureProtection
* ConfigureSsrcs
* ConfigureRids
Bug: None
Change-Id: I013ca438915336d1b8f3477fe8b9f1bf87f514f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138205
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28041}
This CL updates the video packet overhead calculation to make it more
clear. This prepares for future work on improving the accuracy of the
calculation.
Bug: webrtc:9883
Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28040}
The state this set tracks (whether this is new feedback for a packet
belonging to a media ssrc) can already be inferred from data provided
by the SendTimeHistory: if packet send time is not populated in the
feedback it's either because:
1. The feedback has already been processed
2. The receiver is sending feedback for bogus non-existent packets
If the first case, this maps to |feedback_packet_seq_num_set_|
containing the packet, if the ssrc (present in the feedback) is part
of the media ssrcs.
In the second case, this data should be ignored anyway.
Bug: webrtc:10604
Change-Id: If4828091142d68baa8dbb62be9d0b24ccaaa9546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135163
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27882}
If the receiver has indicated that a packet has been received, via a
TransportFeedback RTCP message, it is safe to remove it from the
RtpPacketHistory as we can be sure it won't be needed anymore.
This will reduce memory usage, reduce the risk of overflow in the
history at very high bitrates, and hopefully make payload based padding
a little more useful.
This is code stems partly from
https://webrtc-review.googlesource.com/c/src/+/134208
but without the RtpPacketHistory changes which were landed in
https://webrtc-review.googlesource.com/c/src/+/134307
Bug: webrtc:8975
Change-Id: Iea9d3d32bee5512473744e9ef3a18018567fc272
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135160
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27868}
This reverts commit 3890e99b705065dbc60e6d16932d8584bd67200d.
Reason for revert: Seems to be causing unexpected perf regressions.
Original change's description:
> Remove packets from RtpPacketHistory if acked via TransportFeedback
>
> If the receiver has indicated that a packet has been received, via a
> TransportFeedback RTCP message, it is safe to remove it from the
> RtpPacketHistory as we can be sure it won't be needed anymore.
> This will reduce memory usage, reduce the risk of overflow in the
> history at very high bitrates, and hopefully make payload based padding
> a little more useful.
>
> Bug: webrtc:8975
> Change-Id: I703a353252943f63d7d6edda68f03bc482633fd6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133028
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27745}
TBR=danilchap@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I68ea6cf5c8988d4b625f14a1a9bc556c06a39368
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134161
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27752}
If the receiver has indicated that a packet has been received, via a
TransportFeedback RTCP message, it is safe to remove it from the
RtpPacketHistory as we can be sure it won't be needed anymore.
This will reduce memory usage, reduce the risk of overflow in the
history at very high bitrates, and hopefully make payload based padding
a little more useful.
Bug: webrtc:8975
Change-Id: I703a353252943f63d7d6edda68f03bc482633fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133028
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27745}
Add a version of RTPSenderVideo::GetSentRtpPacketInfo() that operates
over a set of numbers, so as to only grab the lock once.
Bug: webrtc:10501
Change-Id: I9453b0cb44dcd6e2ce196390b2c5c9a7dd6d800a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132014
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27544}
When a LossNotification RTCP message is received, the sequence numbers
it refers to must be converted to timestamps before passing the message
down to the encoder. This CL gives VideoSendStreamImpl access to that
information via VideoSendStreamImpl::rtp_video_sender_.
TBR=sprang@webrtc.org
Bug: webrtc:10501
Change-Id: If207f0b6d2fb344da35b525cc104e8ba5cc614ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131323
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27489}
This reverts commit ab65d8aab5fe63619033371fca1ce2711c2c2137.
Reason for revert: Fails video_engine_tests ExtendedReportsEndToEndTest.TestExtendedReportsCanSignalZeroTargetBitrate
https://ci.chromium.org/p/webrtc/builders/ci/Linux%20MSan/18366
Original change's description:
> Fix target bitrate RTCP messages behavior for SVC streams
>
> Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
> were created. The RTCP target bitrate messages were treated as simulcast
> and were split and send for each separate spatial layer in a separate SSRC.
>
> To fix that an svc flag is now wired to VideoSendStream config
> and filled based on the encoder config in WebrtcVideoEngine. This flag is
> used to differentiate between simulcast and SVC mode in RtpVideoSender.
>
> Bug: webrtc:10485
> Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27345}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: I184f87289d9dccc67de165038d76a5690158a3b5
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130467
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27355}
Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
were created. The RTCP target bitrate messages were treated as simulcast
and were split and send for each separate spatial layer in a separate SSRC.
To fix that an svc flag is now wired to VideoSendStream config
and filled based on the encoder config in WebrtcVideoEngine. This flag is
used to differentiate between simulcast and SVC mode in RtpVideoSender.
Bug: webrtc:10485
Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27345}
CNAME is only set on the first simulcast layer.
It should be set on all of the layers.
Bug: webrtc:10383
Change-Id: Iea345a100769f45d09078adb93e51b7702326492
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126541
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27134}
This is not used in practice as there's functionality on
other levels that serves the same purpose.
Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
to clearly signal passed ownership.
Drop support for accepting nullptr clock to avoid copying the Configuration structure.
Update all calls in webrtc to the new factory function
Bug: None
Change-Id: Ic5a78da8e59ba3988a757a9d9634fa31499ce0db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125901
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26994}
