The purpose of these new methods are to allow creating a RTP packet with
sequence numbers that
can be inspected and is ensured to be sent if SendPacket is invoked.
virtual bool CanSendPacket(const RtpPacketToSend& packet) const = 0;
virtual void AssignSequenceNumber(RtpPacketToSend& packet) = 0;
virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& pacing_info) = 0;
Bug: webrtc:15368
Change-Id: I671e737575e15328e796aa98761a4d540c5812d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343785
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41951}
VoipCore still use RtpSenderEgress::NonPacedPacketSender, therefore
packets sent using NonPacedPacketSender::EnqueuePackets are proxied
to the worker thead.
When NonPacedPacketSender is used, the Pacer already guarantee that packets are sent on the worker queue.
Lock is removed from RtpSenderEgress and instead calls must be made on
the worker thread.
Bug: webrtc:15209
Change-Id: Iaf03377ad8a037ecedbbe588a4c1e8e4eadacd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306960
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40252}
RtpRtcpInterface::RTT follows discouraged style of using return values,
uses raw integers to represent time delta,
and returns values that no code uses (min, max, average RTT)
added LastRtt function addresses all these stylistic issues.
Bug: webrtc:13757
Change-Id: Iaf947dd1b7139026f2beb991e69634c606c6b608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304520
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40028}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
This CL prepares for send packet batching support in later CLs.
Bug: chromium:1439830
Change-Id: I0bbecfa895aa6d4317ef8049b3789272a440d032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304282
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40009}
The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.
Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
The CSRC concept is really a frame level concept.
Setting it per sender is a quick hack, and should be minimized.
This function doesn't seem to be used anywhere, so removing it
lessens the chance of confusion.
Bug: webrtc:7135
Change-Id: Ia3c27b5984b153e68bc51d93b03f08f7f867adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286426
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#38822}
In some upcoming use cases we might wish to flush pending
retransmissions from the pacer queue. In order to not make those packets
forever inaccessible this CL adds a way to clear their "pending" status
from the packet history.
Bug: webrtc:11340
Change-Id: I9aac44125899a7f1e5a5e5be3390ac07b1e661ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38037}
This reverts commit 2c41cbae37.
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 fb0dca6c05.
>
> 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 fb0dca6c05.
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}
The config flag will be removed once downstream usage is gone.
Bug: webrtc:11340
Change-Id: Iee8816660009211540d9b09bb3cba514455d709b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228431
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34757}
Since https://webrtc-review.googlesource.com/c/src/+/228433 both audio
and video now only call Get/SetRtpState while not registered to the
packet router.
We can thus remove the lock around packet sequencer and just use a
thread checker.
Bug: webrtc:11340
Change-Id: Ie6865cc96c36208700c31a75747ff4dd992ce68d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228435
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34755}
With deferred packet sequencing, the PacketSequencer instance is called
directly from the RtpRtcp module while before it was called from within
the RTPSender while holding a lock.
Since sequence number assignment happens on the same thread as actual
packet sending, though thought was that locking was no longer needed.
Unfortunately, SetRtpState()/GetRtpState() also exists - and while they
should only be called on creating/destruction there is a possible race
where a delayed packet from the pacer accesses the sequencer while
GetRtpState() is being called.
For now, this CL just adds a lock to guard sequencer. Follow-ups will
make sure get/set state is never called while module is attached to
the packet router. After that, the lock can be removed again.
Bug: webrtc:11340, webrtc:12470
Change-Id: I123c762fb4afd20b3a6bd03b86234eb9ec34a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228430
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34723}
With this turned on, packets will be sequence number after the pacing
stage rather that during packetization.
This avoids a race where packets may be sent out of order, and paves
the way for the ability to cull packets from the pacer queue without
causing sequence number gaps.
For now, the feature is off by default. Follow-ups will enable it for
video and audio separately.
Bug: webrtc:11340, webrtc:12470
Change-Id: I6d411d8c85b9047e3e9b05ff4c2c3ed97c579aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208584
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34661}
This prepares for deferred sequence numbering, and is (sort of)
extracted from
https://webrtc-review.googlesource.com/c/src/+/208584
Bug: webrtc:11340, webrtc:12470
Change-Id: I2f3695309e1591b9f7a1ee98556f4f0758de7f69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227352
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34643}
This will allow us to enable receive-side RTT without having to recreate all AudioReceiveStream objects.
Bug: webrtc:12951
Change-Id: I1227297ec4ebeea9ba15fe2ed904349829b2e669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225262
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34464}
This change achieves an Idle Wakeup savings of 200 Hz.
ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.
Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
This change migrates RTCP send polling happening in
ModuleRtpRtcpImpl2::Process to task queues.
ModuleRtpRtcpImpl2 would previously only cause RTCP sends while being
registered with a ProcessThread. This is now relaxed so that RTCP will
be sent regardless of ProcessThread registration status, and it seems
no tests cared.
Now there's only one piece of polling left in Process.
Bug: webrtc:11581
Change-Id: Ibdcffefccef7363f2089c34a9c7d694d222445c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222603
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34350}
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats`
via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams.
`GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and
`ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`.
Bug: webrtc:12529
Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33452}
`LastReceivedNTP()` does not need to be part of the public members of
`ModuleRtpRtcpImpl` and `ModuleRtpRtcpImpl2` since it is used only
once in the same class.
This change is requried by the child CL [1] which adds a public getter
needed to add remote-outbound stats.
[1] https://webrtc-review.googlesource.com/c/src/+/211041
Bug: webrtc:12529
Change-Id: I82cfea5ee795de37fffa3d759ce9f581ca775d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211043
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33420}
This reverts commit f23e2144e8.
Reason for revert: Need further discussion on appropriate thread/tq requirements.
Original change's description:
> Add task queue to RtpRtcpInterface::Configuration.
>
> Let ModuleRtpRtcpImpl2 use the configured value instead of
> TaskQueueBase::Current().
>
> Intention is to allow construction of RtpRtcpImpl2 on any thread.
> If a task queue is provided (required for periodic rtt updates), the
> destruction of the object must be done on that same task queue.
>
> Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
>
> Bug: None
> Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32949}
TBR=danilchap@webrtc.org,ilnik@webrtc.org,saza@webrtc.org,nisse@webrtc.org,srte@webrtc.org
Change-Id: I7e5007f524a39a6552973ec9744cd04c13162432
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201420
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32953}
Let ModuleRtpRtcpImpl2 use the configured value instead of
TaskQueueBase::Current().
Intention is to allow construction of RtpRtcpImpl2 on any thread.
If a task queue is provided (required for periodic rtt updates), the
destruction of the object must be done on that same task queue.
Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
Bug: None
Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32949}
Methods deleted: StorePackets, RtcpXrRrtrStatus. They are now private
methods on the two implementations.
Bug: None
Change-Id: If68e8f1e8ba233302e24e0cdb6bf7c1b0c9f330f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194322
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32670}
This is a reland of f5e261aaf6
This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
This reverts commit f5e261aaf6.
Reason for revert: Breaks downstream projects.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.
The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.
Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
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}
