This reverts commit 66147e892dd6b7b1beaddbcab456a1ce28b2ad22.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Optimize PacketRouter/RTPSender interactions."
>
> This reverts commit 6f129b3b7605dc69c8c188ca02d133250130570e.
>
> Reason for revert: Speculative revert (some perf test are failing)
>
> Original change's description:
> > Optimize PacketRouter/RTPSender interactions.
> >
> > The legacy code-path uses a hashmap as cache in order to speed up
> > finding the right rtp module to send on. The new path should use that
> > as well.
> > In addition, there are checks that verify if an RTP module can send
> > padding, in some cases payload based. These result in a number of
> > calls to methods in RTPSender requiring its lock to be taken. This CL
> > introduces a combined SupportsPadding() check method which performs
> > all those checks in one go.
> >
> > Bug: None
> > Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28535}
>
> TBR=asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
>
> Change-Id: I8499dc0fd6e6d0b9fa7a0886c8754655e5589780
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145326
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28552}
TBR=mbonadei@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I3bff3ecb2b776e30f77c1884f6faa72b21788017
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145401
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28563}
This reverts commit 8b3e4e2d1166464f6b309f4fc533a29607d2771f.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Reland "Add ability to set RTCP sender ssrc at construction time""
>
> This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
>
> Reason for revert: Speculative revert (some perf test are failing)
>
> Original change's description:
> > Reland "Add ability to set RTCP sender ssrc at construction time"
> >
> > This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
> >
> > Patch set 1 is the original CL.
> > Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> > if either current SSRC is 0 or if the SSRC is identical to the current
> > one. If so, don't schedule an early report.
> > This prevents a regression in which audio jitter became too low(?)
> >
> > Original change's description:
> > > Add ability to set RTCP sender ssrc at construction time
> > >
> > > Bug: webrtc:10774
> > > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28506}
> >
> > Bug: webrtc:10774
> > Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28520}
>
> TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10774
> Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28555}
TBR=mbonadei@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
Change-Id: I2e5c17e8edfd938424f901222158643baa04866e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28562}
This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Reland "Add ability to set RTCP sender ssrc at construction time"
>
> This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
>
> Patch set 1 is the original CL.
> Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> if either current SSRC is 0 or if the SSRC is identical to the current
> one. If so, don't schedule an early report.
> This prevents a regression in which audio jitter became too low(?)
>
> Original change's description:
> > Add ability to set RTCP sender ssrc at construction time
> >
> > Bug: webrtc:10774
> > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28506}
>
> Bug: webrtc:10774
> Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28520}
TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10774
Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28555}
This reverts commit 6f129b3b7605dc69c8c188ca02d133250130570e.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Optimize PacketRouter/RTPSender interactions.
>
> The legacy code-path uses a hashmap as cache in order to speed up
> finding the right rtp module to send on. The new path should use that
> as well.
> In addition, there are checks that verify if an RTP module can send
> padding, in some cases payload based. These result in a number of
> calls to methods in RTPSender requiring its lock to be taken. This CL
> introduces a combined SupportsPadding() check method which performs
> all those checks in one go.
>
> Bug: None
> Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28535}
TBR=asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I8499dc0fd6e6d0b9fa7a0886c8754655e5589780
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145326
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28552}
This reverts commit bb7727211c535f8a9dce27891941b52b6ea8e750.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Make new pacer padding more like old one
>
> The (currently unused) new pacer code path was implemented with what
> was intended as a more careful padding strategy.
> Unfortunately this doesn't work as well as expected due to the fact
> that the padding budget cannot build up underuse.
>
> I made another CL that could fix that, but I think it adds complexity
> for dubious gains. It also will make it more difficult to find any
> potential regression when switching to the new path, should one occur.
> See https://webrtc-review.googlesource.com/c/src/+/144563
>
> Therefore, this CL makes the new code path choose RTX payload in the
> same way as is currently done.
>
> Bug: webrtc:10633
> Change-Id: If2115d4fa7463add959faa77c63101286c27e0f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145202
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28537}
TBR=sprang@webrtc.org,stefan@webrtc.org
Change-Id: I99b72858414e0a245da596d94694449da88fd626
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145324
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28550}
This reverts commit 7325bc3917e6dd4c92e7a18fd879ba91f0b2851f.
Reason for revert: FecTest.UlpfecTest is consistently failing.
Original change's description:
> Refactor FEC code to use COW buffers
>
> This refactoring helps to reduce unnecessary memcpy calls on the receive
> side.
>
> This CL is the first stage of refactoring: it only replaces
> |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| and does
> necessary changes.
>
> A follow-up CL will remove length field of the Packet class.
>
>
> Bug: webrtc:10750
> Change-Id: Ie233da83ff33f6370f511955e4c65d59522389a7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144881
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28539}
TBR=brandtr@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Change-Id: I07c34256a76174f09a0d27eacbae6488e66f4b43
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145340
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28545}
This refactoring helps to reduce unnecessary memcpy calls on the receive
side.
This CL is the first stage of refactoring: it only replaces
|uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| and does
necessary changes.
A follow-up CL will remove length field of the Packet class.
Bug: webrtc:10750
Change-Id: Ie233da83ff33f6370f511955e4c65d59522389a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144881
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28539}
The (currently unused) new pacer code path was implemented with what
was intended as a more careful padding strategy.
Unfortunately this doesn't work as well as expected due to the fact
that the padding budget cannot build up underuse.
I made another CL that could fix that, but I think it adds complexity
for dubious gains. It also will make it more difficult to find any
potential regression when switching to the new path, should one occur.
See https://webrtc-review.googlesource.com/c/src/+/144563
Therefore, this CL makes the new code path choose RTX payload in the
same way as is currently done.
Bug: webrtc:10633
Change-Id: If2115d4fa7463add959faa77c63101286c27e0f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145202
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28537}
The legacy code-path uses a hashmap as cache in order to speed up
finding the right rtp module to send on. The new path should use that
as well.
In addition, there are checks that verify if an RTP module can send
padding, in some cases payload based. These result in a number of
calls to methods in RTPSender requiring its lock to be taken. This CL
introduces a combined SupportsPadding() check method which performs
all those checks in one go.
Bug: None
Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28535}
Removing extension will be used in DatagramDtlsAdaptor to remove transport sequence number to avoid having both datagram and RTP feedback loops. The sequence number will be stored in temporary map and used to re-create RTCP fdeedback packed when we receive datagram ACK. It would enable integration of Datagram transport without any changes in the upper layers of RTP stack. Note that Datagram adaptor changes will be implemented in a separate changelist.
In this change:
- Implement method to remove extension by rebuilding entire packet without given extension type.
- Fails if extension is not registered or not set.
- Unit test
Bug: webrtc:9719
Change-Id: I9d3811d5d97fadde5a294d5da643b2ebc6a8196e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28530}
This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
Patch set 1 is the original CL.
Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
if either current SSRC is 0 or if the SSRC is identical to the current
one. If so, don't schedule an early report.
This prevents a regression in which audio jitter became too low(?)
Original change's description:
> Add ability to set RTCP sender ssrc at construction time
>
> Bug: webrtc:10774
> Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28506}
Bug: webrtc:10774
Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28520}
This CL makes the new code path for paced sending functionally complete.
By default, the field trial WebRTC-Pacer-ReferencePackets is Enabled,
meaning that there is no behavior change unless the field trial is
forced to Disabled. This is done in tests, and can be done on the
command line for manual testing.
Bug: webrtc:10633
Change-Id: I0d66c94ef83b5847dee437a785018f09ba3f828d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144050
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28497}
This CL adds a number of debug-mode checks for inconsistent state, and
if in release mode will reset the history instead of crashing.
Bug: webrtc:10794
Change-Id: If099a1bb61314177cdad633d0fbdca052cd3a5ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144525
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28475}
This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at:
http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
We are still missing the code to:
- Negotiate the header extension.
- Collect capture time for audio and video and have the info sent with the header extension.
- Receive the header extension and use its info.
Bug: webrtc:10739
Change-Id: I75af492e994367f45a5bdc110af199900327b126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28468}
This is a small utility method to check whether an extension has been
reserved, so that can be checked before attempting to set an extension
without the need to actually try setting it and potentially failing
with warning loggins as a result.
Bug: webrtc:10633
Change-Id: Ie6f2c4f3f5e94a30dbf60aec6290ebee72681d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144461
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28455}
Unlike TimeToSendPadding(), the new GeneratePadding() method will
generate RTP packets and put them in the pacer queue, which will be
responsible for actually sending them.
A slight difference from previous logic is that we do not use a lower
bound of 50bytes for getting payload packets, instead we always try and
then abort if the next padding packet is larger than the current
available budget.
Since we're not sending the packets immediately, we don't need to worry
about twcc sequence numbering or updating the stats, that will be
handled by the general SendPacket() codepath. We can also omit the
PacingInfo struct and the return value of bytes sent, as that will
be handled when taking the packets out of the queue.
Bug: webrtc:10633
Change-Id: I066c292805a0bf76c59f68e66c21ea23fdb56c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143794
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28403}
This CL allows for FEC protection of packets with VideoTimingExtension by
zero-ing out data, which is changed after FEC protection is generated (i.e
in the pacer or by the SFU).
Actual FEC protection of these packets would be enabled later, when all
modern receivers have this change.
Bug: webrtc:10750
Change-Id: If4785392204d68cb8527629727b5c062f9fb6600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143760
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28396}
This CL also improves test coverage and fixes an issue where the
(until now) unused code path for useful padding did not respect the
lower bound packet sizes.
Bug: webrtc:8975
Change-Id: I065745ca7ac9f7098d796c6a015cd96f052ee94f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142801
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28343}
This method will be called when PacedSender is using the new code path
that directly owns the packets to be sent.
It can be seen as combining a few features of the old code path:
* It checks if this is the correct RTP module and then sends, without
the need for PacketRouter to poll multiple methods for SSRC etc first.
* It partly corresponds to TimeToSendPacket(), but RTX encapsulation
now happens pre-pacer and FEC does not need to have a packet history,
so most of that method is not used.
* It implements most of PrepareAndSendPacket(), such as updating header
extensions, reporting stats and of course forwards to Transport. It
now also handles the history a bit differently, since media packets
will only be stored for potential retransmission post-pacer.
Bug: webrtc:10633
Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28298}
This CL just adds the new interfaces, follow-ups will add implementation
in various parts of the code, and then do cleanup once usage of old
interface is gone.
Bug: webrtc:10633
Change-Id: Icd916f4220065c0d0e4f3f0bfaaed248f8c70d08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140891
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28252}
This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources.
Bug: webrtc:10545 webrtc:10668
Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28249}
This method will be used instead of GetPacketAndSetSendTime() when the
new pacer code path is used, where the packet isn't stored in the
history during pacing.
Bug: webrtc:10633
Change-Id: Ie168125d949cef617ade3868a1858ed1dffe909c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140892
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28240}
These are replaced with the methods SendPictureLossIndication and
SendFullIntraRequest, added in cl
https://webrtc-review.googlesource.com/c/src/+/140043.
Also delete the corresponding state variable
RtpRtcpImpl::key_frame_req_method_, the enum KeyFrameRequestMethod,
and the nearby unused enum RtpRtcpPacketType.
Bug: None
Change-Id: I1ac2e4ce6dbe20d1d1cbb3d5b2256ea55b341a57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28221}
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}
Replaced by separate methods
SendPictureLossIndication and SendFullIntraRequest.
The split SetKeyFrameRequestMethod/RequestKeyFrame implicitly
requires that the two methods are called on the same thread, to avoid a
data race. After downstream code is updated, both deprecated
methods and the member |ModuleRtpRtcpImpl::key_frame_req_method_| can
be deleted.
Bug: None
Change-Id: I454f6d16b667f2306cba0dec467ddc183ad449c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140043
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28163}
Currently, the packet in the history that most closely matches the bit
budget between two PacedSender::Process() calls is selected to be
retransmitted. This usually means that the smallest packet in the
history is selected over and over.
With this new field trial, we ignore the size constraint (since you're
sending padding, you obviously have some bandwidth to spare) and
instead prefer packets that have the fewest transmission times first,
and after that we prefer new packets over older ones. This way, in
case of available bandwidth but small loss, these padding packets have
a greater chance of actually being useful to the receiver.
Bug: webrtc:8975
Change-Id: I15a69231f44bfbefcb9ab38dd7886b966e3af6fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135954
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28084}
The imports of Packet, ReceivedPacket from ForwardErrorCorrection::
collides with other usages of the names introduced in a followup CL.
Bug: webrtc:9883
Change-Id: Ib042c9091ad8e350cbdf01b837af09c820dbff33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138279
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28075}
