Now update the playout delay and related stats on the worker thread.
This was previously reviewed here:
https://webrtc-review.googlesource.com/c/src/+/172929/
With the exception of reducing unnecessarily broad
lock scope in one function in rtp_rtcp_impl.cc
and added comments in rtp_streams_synchronizer.h
Bug: webrtc:11489
Change-Id: I77807b5da2accfe774255d9409542d358f288993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174200
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31193}
This interface has a couple of issues. Primarily for me, it makes it
difficult work with the paced sender as we need to either temporarily
release a lock or force a thread-handover in order to avoid a cyclic
lock order.
For video in particular, its behavior is also falky since header sizes
can vary not only form frame to frame, but from packet to packet within
a frame (e.g. TimingInfo extension is only on the last packet, if set).
On bitrate allocation, the last reported value is picked, leading to
timing issues affecting the bitrate set.
This CL removes the callback interface and instead we simply poll the
RTP module for a packet overhead. This consists of an expected overhead
based on which non-volatile header extensions are registered (so for
instance AbsoluteCaptureTime is disregarded since it's only populated
once per second). The overhead estimation is a little less accurate but
instead simpler and deterministic.
Bug: webrtc:10809
Change-Id: I2c3d3fcca6ad35704c4c1b6b9e0a39227aada1ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173704
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31185}
RtpVideoSender now stores fec type and overhead instead of querying the
generator all the time. Setting of protection parameters and asking for
current bitrate is also now handled just by the VideoFecGenerator
instance, instead of going via RtpVideoSender.
Finally, adds method to query for RtpState in VideoFecGenerator
interface. This avoids an ugly cast that would have been even more
trouble after moving fec generation.
For context, see https://webrtc-review.googlesource.com/c/src/+/173708
Bug: webrtc:11340
Change-Id: Ia5e6cd919e71850c9cc5ed5a4f4417338d577162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174203
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31166}
Add MockFrameTransformer to test/, and remove definitions from unit test
files.
Bug: webrtc:11380
Change-Id: Ia709883e8d000852e3f71e7bfb87877072e22aeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31151}
This CL also slightly refactors unit test, to test fewer things each.
Bug: webrtc:11508
Change-Id: I98553d2b185364132c626d373683f38891e36c6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31087}
The transformer was previously moved into the config of the first stream
which resulted in incorrect behavior for simulcast. Use the transformer
in all the streams.
Pass the sender's ssrc on registring the transformed frame callback, to
associate separate transformer sinks for each sender.
Bug: chromium:1065838
Change-Id: I5c52dacb241c68268681b85f875257b24987849e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173332
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31050}
Instead dependency descriptor can be used to communicate discardability
Bug: webrtc:11358
Change-Id: I46b4f551acd002d4355d18033e03d8181ec94c6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172922
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31004}
Add a new frame interface to be used by frame transformers in Insertable
Streams. TransformableFrameInterface will replace
video_coding::EncodedFrame in a follow up CL, once downstream
dependecies are updated to use the new interface.
Until the functions using video_coding::EncodedFrame are removed from
the API, the video sender and receiver frame transformer delegates call
both function versions to avoid breaking tests downstream.
The TransformableFrameInterface will be used for both audio and video
frame transformers in follow-up CLs.
Bug: webrtc:11380
Change-Id: I9389a8549c156e13b1d8c938ff51eaa69c502f33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171863
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30941}
This allows trading off some potential media quality for CPU usage.
Bug: webrtc:8975
Change-Id: I447a03f596e9e711ba5d7038fe71f27bd80bf795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172085
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30936}
As far as I can tell, every call site already populates this field, so
we can now remove it.
Bug: webrtc:8975
Change-Id: I58515dd16d4ad8bf8872077b67a67f6e92e7b542
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30857}
make it a member function which allows to reduce number of parameters
and simplify accessing more state in the future.
Bug: None
Change-Id: Iba35125c0c2cf1d6bb67b106c1f73a33ecb8e44e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170366
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30797}
update MOCK_METHODs to use new syntax recommended in go/totw/164
Replace fixture with struct of mocks.
Use main method under test (IncomingPacket) directly rather than through fixture helpers
minor cleanup of the RtcReceiver itself:
make IncomingPacket function more friendly to containers,
mark class as final to verify ability to inherit from it is not used and
thus destructor doesn't need to be virtual.
Bug: None
Change-Id: I346e7dc513b1fbe663ebe5858dec7df0520416a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170226
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30765}
With the new config option |always_send_mid_and_rid|, the user
of the RTPSender can decide if MIDs and RIDs should always be sent
(when provided and negotiated), or if their sending should be disabled
after the receiver has acked the stream. Depending on the use case,
different settings might be preferable. The default setting is not
changed in this CL.
Bug: webrtc:11416
Change-Id: Ic3c71a6105fb7ee08d53cf9eb03f4e53b5799610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170109
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30742}
This is a reland of 49734dc0faa69616a58a1a95c7fc61a4610793cf
Patchset 2 contains a fix for the fuzzer set up. Since we now parse
an RtpPacket out of the fuzzer data, the header needs to be correct,
otherwise we fail before even reaching the FEC code that we actually
want to test.
Bug: webrtc:11340, chromium:1052323, chromium:1055974
TBR=stefan@webrtc.org
Original change's description:
> Reland "Refactors UlpFec and FlexFec to use a common interface."
>
> This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
>
> Original change's description:
> > Refactors UlpFec and FlexFec to use a common interface.
> >
> > The new VideoFecGenerator is now injected into RtpSenderVideo,
> > and generalizes the usage.
> > This also prepares for being able to genera FEC in the RTP egress
> > module.
> >
> > Bug: webrtc:11340
> > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30515}
>
> Bug: webrtc:11340, chromium:1052323
> Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30593}
Bug: webrtc:11340, chromium:1052323
Change-Id: Ib8925f44e2edfcfeadc95c845c3bfc23822604ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169222
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30724}
This changes removes an extra layer of indirection
since RtcpTransceiver doesn't own TaskQueue it uses.
Bug: None
Change-Id: Ie1ef4cd8c3fb18a8e0b7ddaf0d6a319392b9e9f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126040
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30704}
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}
This cl add a configuration flag to allow REMB messages to be sent immediately when the bitrate value have changed.
The remb message is still included in all following compound packets.
Bug: None
Change-Id: I9f71d30cddbccd095e1d2971247c731bd1727d32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169221
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30627}
This reverts commit 49734dc0faa69616a58a1a95c7fc61a4610793cf.
Reason for revert: Still something wrong with ulpfec fuzzer setup.
Original change's description:
> Reland "Refactors UlpFec and FlexFec to use a common interface."
>
> This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
>
> Original change's description:
> > Refactors UlpFec and FlexFec to use a common interface.
> >
> > The new VideoFecGenerator is now injected into RtpSenderVideo,
> > and generalizes the usage.
> > This also prepares for being able to genera FEC in the RTP egress
> > module.
> >
> > Bug: webrtc:11340
> > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30515}
>
> Bug: webrtc:11340, chromium:1052323
> Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30593}
TBR=sprang@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11340, chromium:1052323
Change-Id: I920ce0a48a08768d7a98a563e2b66bd6eb8602b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169121
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30616}
This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
Original change's description:
> Refactors UlpFec and FlexFec to use a common interface.
>
> The new VideoFecGenerator is now injected into RtpSenderVideo,
> and generalizes the usage.
> This also prepares for being able to genera FEC in the RTP egress
> module.
>
> Bug: webrtc:11340
> Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30515}
Bug: webrtc:11340, chromium:1052323
Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30593}
same way as generic frame descriptor is authenticated.
Bug: webrtc:10342
Change-Id: I50bb3ab343d66f1f628083183444da6e338f7db9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168681
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30578}
The change ships GenericDescriptor00 and authentication by default,
but doesn't expose it by default, and makes WebRTC respond to
offers carrying it.
The change adds a unit test for the new semantics.
Tests well in munge-sdp. Frame marking replaced by
http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00
in the offer results in an answer containing the
extension as first entry.
Bug: webrtc:11367
Change-Id: I0ef91b7d4096d949c3d547ece7d6c4d39aa241da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168661
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30542}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
with adjustments.
Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
This reverts commit 11af1d7444fd7438766b7bc52cbd64752d72e32e.
Reason for revert: Possible crash
Original change's description:
> Refactors UlpFec and FlexFec to use a common interface.
>
> The new VideoFecGenerator is now injected into RtpSenderVideo,
> and generalizes the usage.
> This also prepares for being able to genera FEC in the RTP egress
> module.
>
> Bug: webrtc:11340
> Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30515}
TBR=brandtr@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Iddf112d801621c8a4370b853cee3fa42bf2c7fba
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30524}
The new VideoFecGenerator is now injected into RtpSenderVideo,
and generalizes the usage.
This also prepares for being able to genera FEC in the RTP egress
module.
Bug: webrtc:11340
Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30515}
This reverts commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
Reason for revert: breaks downstream tests
Original change's description:
> Do not propagate generic descriptor on receiving frame
>
> It was used only for the frame decryptor.
> Decryptor needs only raw representation that it can recreate
> in a way compatible with the new version of the descriptor.
>
> Bug: webrtc:10342
> Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30501}
TBR=danilchap@webrtc.org,sprang@webrtc.org,philipel@webrtc.org
Change-Id: I6634df06ee75aa8cdfda614994ab11f7a5845c70
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168488
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30502}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
Bug: webrtc:10342
Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30501}
References to PlayoutDelayOracle and the deprecated RtpSenderVideo
constructor have been removed in downstream code, we can now clean the
unused code away.
Bug: webrtc:10809, webrtc:11340
Change-Id: I789274be2079ad4ddd7e83a5fa249b06a32a4e82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168400
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30499}
This is a reland of 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2
Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
>
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
>
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
>
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
>
> This allows containing the logic fully within RTPSenderVideo.
>
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}
TBR=stefan@webrtc.org
Bug: webrtc:11340
Change-Id: I2fdd0004121b13b96497b21e052359e31d0c477a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168305
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30479}
This is in preparation of an upcoming CL that will propagate this
information through the TransportFeedbackAdapter.
Bug: webrtc:10932
Change-Id: Ic2a026b5ef72d6bf01e698e7634864fedc659b4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168220
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30476}
This reverts commit 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2.
Reason for revert: Breaks downstream project
Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
>
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
>
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
>
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
>
> This allows containing the logic fully within RTPSenderVideo.
>
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ide922e680ae36bb69b95e58002482cf5ed57e254
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30475}
The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
header extension was successfully propagated to the receiving side. Once
it was determined that the receiver had received a frame with the new
delay tag, it's no longer necessary to propagate.
The issue with this implementation is that it is based on max
extended sequence number reported via RTCP, which makes it often slow
to react, could theoretically fail to produce desired outcome (max
received > X does not guarantee X was fully received and decoded), and
added a lot of code complexity.
The guarantee of delivery can in fact be accomplished more reliably and
with less code by making sure to tag each frame until an undiscardable
frame is sent.
This allows containing the logic fully within RTPSenderVideo.
Bug: webrtc:11340
Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30473}
