This prepares for ability to defer sequence number assignment to after
the pacing stage - a scenario where the RtpRtcp module rather than than
RTPSender class has responsibility for sequence numbering.
Bug: webrtc:11340
Change-Id: Ife88f60258b9b7cfd9dbd3326f02ac34da8f7603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214967
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33702}
These functions are not longer used by the RtpRtcp implementations.
Bug: None
Change-Id: Ibc36433b253b264de4cdcdf380f5ec1df201b17a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207862
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33282}
When calculating maximum allowed size for a media packet.
In particular take in account that rtx packet might need to
include mid and repaired-rsid extensions when media packet can omit them.
Bug: webrtc:11031
Change-Id: I3e7bc36437c23e0330316588d2a46978407c8c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206060
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33184}
This CL:
1) Updates RtpSenderVideo to actually populate the is_key_frame field
properly.
2) Updates UlpfecGenerator to:
* Allow updating the protection parameters before adding any packet.
* Apply keyframe protection parameter when at least one buffered
media packet to be protected belongs to a keyframe.
Updating the parameters in the middle of a frame is allowed, at that
point they only determine how many _complete_ frames are needed in order
to trigger FEC generation. Only that requirement is met, will the
protection parameters (e.g. FEC rate and mask type) actually be applied.
This means that delta-frames adjecent to a key-frame (either ahead of
or after) may be protected in the same way as the key-frame itself.
Bug: webrtc:11340
Change-Id: Ieb84d0ae46de01c17b4ef72251a4cb37814569da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32787}
This CL fixes a bug where the RtpPackeToSend::fec_protect_packet flag
was not cleared when a packet copy was fetched from the packet history
in order to be retransmitted. This caused the packet to be added to the
FEC generator a second time when the retransmission passed through
RtpSenderEgress.
The bug did not affect RTX retransmission and only manifests when using
deferred FEC generation.
Bug: webrtc:11340
Change-Id: Ic7ce2800cce9a99e74bd3dd697bc0779d2a02fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185817
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32227}
It is not longer needed by the rtp_rtcp module.
Bug: webrtc:6471
Change-Id: I89a4374a50c54a02e9f20a5ce789eac308aaffeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179523
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31773}
Some classes such as RtpSenderEgress makes assumptions about which
threads (e.g. paced sender vs worker thread) call specific methods.
Unit tests currently are single threaded so these checks are
essentially noops.
This change uses a separate task queue for calls epected to be called
by the pacer, so that inconsistencies in thread can be detected early.
Bug: None
Change-Id: Ic0904304a67eb034033524e62306da34b9eab8b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178602
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31628}
This is a reland of 19df870d92
Patchset 1 is the original.
Subsequent patchset changes threadchecker that crashed with downstream
code.
Original change's description:
> Reland "Allows FEC generation after pacer step."
>
> This is a reland of 75fd127640
>
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
>
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
>
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}
Bug: webrtc:11340
Change-Id: Ib741c8c284f523c959f8aca454088d9eee7b17f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178600
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31619}
This reverts commit 19df870d92.
Reason for revert: Downstream project failure
Original change's description:
> Reland "Allows FEC generation after pacer step."
>
> This is a reland of 75fd127640
>
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
>
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
>
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: I3b2b25898ce88b64c2322f68ef83f9f86ac2edb0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178563
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31614}
This is a reland of 75fd127640
Patchset 2 contains a fix. Old code can in factor call
RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
is not supported there - we shouldn't crash.
Original change's description:
> Allows FEC generation after pacer step.
>
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
>
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
>
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
>
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}
Bug: webrtc:11340
Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31613}
On the way remove need for lock for
rtp_sequence_number_map_ and timestamp_offset_.
Change-Id: I21a5cbf6208620435a1a16fff68c33c0cb84f51d
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177424
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31581}
The timer fired a Notify call that goes to an object that already
receives callbacks for every packet from RtpSenderEgress.
Further optimizations will be realized by moving ownership
of the stats to the worker thread and then be able to remove
locking in a few classes that currently are tied to those
variables and the callbacks that previously did not come
from the same thread consistently.
We could furthermore get rid of one of these callback interfaces
and just use one.
Bug: webrtc:11581
Change-Id: I56ca5893c0153a87a4cbbe87d7741c39f9e66e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177422
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31575}
This reverts commit 75fd127640.
Reason for revert: Breaks downstream test
Original change's description:
> Allows FEC generation after pacer step.
>
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
>
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
>
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
>
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: Ie714e5f68580cbd57560e086c9dc7292a052de5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177983
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31559}
Split out from https://webrtc-review.googlesource.com/c/src/+/173708
This CL enables FEC packets to be generated as media packets are sent,
rather than generated, i.e. media packets are inserted into the fec
generator after the pacing stage rather than at packetization time.
This may have some small impact of performance. FEC packets are
typically only generated when a new packet with a marker bit is added,
which means FEC packets protecting a frame will now be sent after all
of the media packets, rather than (potentially) interleaved with them.
Therefore this feature is currently behind a flag so we can examine the
impact. Once we are comfortable with the behavior we'll make it default
and remove the old code.
Note that this change does not include the "protect all header
extensions" part of the original CL - that will be a follow-up.
Bug: webrtc:11340
Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31558}
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.
Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.
The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.
Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
In https://webrtc-review.googlesource.com/c/src/+/173704 the overhead
calculations were made more static, so that "volatile" extensions
(those that are not set on every packet) are ignored. The intent, as
the comments specify, was to ignore RepairedRtpStreamId since that is
only used on RTX packets.
This CL makes us actually count that extension as volatile.
Bug: webrtc:10809
Change-Id: If42ae84e4c09ff9112e93f8d872ee890c6253a23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175010
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31246}
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}
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}
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}
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 49734dc0fa
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 11af1d7444
>
> 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 reverts commit 49734dc0fa.
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 11af1d7444
>
> 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 11af1d7444
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}
This reverts commit 11af1d7444.
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 is a reland of 4f68f5398d
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 4f68f5398d.
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}
* Removes legacy defines from rtp_rtcp_defines.
* Simplifies the feedback adaptation logic, this is achieved
by using the ability to preserve lost packets information
from the RTCP message.
* Extracts in flight data tracking to a separate helper class.
* Removes legacy fields and constructors from the PacketFeedback
structure.
* Removes the legacy GetTransportFeedbackVector method.
Apart from reducing total LOC, this prepares for moving the adaptation
to run on a TaskQueue.
Bug: webrtc:9883
Change-Id: I5ef4eace0948f119f283cd71dc2b8d0954a1449b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29674}
All plumbing was landed a while ago, but this call site was not updated.
This change aims to reduce contention/overhead when posting large
number of packets to the paced sender.
Bug: webrtc:10809
Change-Id: I5486131b980e55331a38151bceee1cb96e35a942
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158203
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29599}
Post-pacer code now contained in RtpSenderEgress class.
For now, this is a member of RTPSender. More refactoring is needed to
make clean split.
Bug: webrtc:11036
Change-Id: I95264d013de120601784f130ba81c7b234446980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157172
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29519}
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 17608dc459
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 is a reland of 17608dc459
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}
