The methods are no longer in use, this CL cleans away references and
updates any tests using them.
Bug: webrtc:10633
Change-Id: I2db301e0a021a2f85a8b9a74e409303baba407da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150520
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@{#28956}
And a corresponding struct RtpReceiveStats. This is intended
to hold the information exposed via GetStats, which is quite
different from the stats reported to the peer via RTCP.
This is a preparation for moving ReceiveStatistics out of the
individual receive stream objects, and instead have a shared instance
owned by RtpStreamReceiverController or maybe Call.
Bug: webrtc:10679,chromium:677543
Change-Id: Ibb52ee769516ddc51da109b7f2319405693be5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148982
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28943}
This change adds helper classes to manipulate Absolute Capture Time header extensions. Both classes support the "timestamp interpolation" optimization.
Bug: webrtc:10739
Change-Id: I08eff46eb8910842a6dbaa3288b976004fabe1c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149801
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28936}
This flag has been default-off since Jul 24th (m77 branch) and apart
from a bug fixed on Aug 5th, there have been no reports of issues, so
let's remove it and start cleaning away the old code path.
Most of the usage within RtpSender/PacingController and their
respective unit tests are removed with this CL, but there will be
several more to follow.
Bug: webrtc:10633
Change-Id: I1986ccf093434ac8fbd8d6db82a0bb44f50b514e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149838
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28930}
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.
The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.
Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
There are currently three overloads with different number of arguments,
and one of those return a raw pointer. This cl changes that to unique_ptr.
The transition plan is to update those downstream call sites that
currently require a raw pointer to use one of the other overloads.
Bug: webrtc:10679
Change-Id: I234605e99c04a59fbe6f478581ed8edd96a9b05a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28804}
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.
Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
Cname callback is used only on receive side, and statistics (soon)
only on the send side.
Bug: webrtc:10679
Change-Id: I122e9cafaea93cd0ba75dc955a652d9d4bddc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147867
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@{#28767}
This change removes the old `ContributingSources` class. It has been replaced by the new `SourceTracker`.
Bug: webrtc:10793
Change-Id: Ibd481cf6584837c46b229b9fc2a071362f07d361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147878
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28756}
Old way to produce this histogram was based on RtcpStatisticsCallback
reporting sent RTCP messages, with some additional processing by the
ReportBlockStats class. After this cl, to grand average fraction loss
is computed by StreamStatistician, queried by VideoReceiveStream when
the stream is closed down, and passed to ReceiveStatisticsProxy which
produces histograms.
This is a preparation for deleting the RtcpStatisticsCallback from
ReceiveStatistics.
Bug: webrtc:10679
Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28747}
This is a race that can happen if a nack arrives before media is
disabled, but the packet is not processed until after the disabling
is complete.
Bug: webrtc:10633, b/138636698
Change-Id: Ic90462b815163ab58c324e5cdb95c8d199c0b772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147277
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28718}
That method will be retired, but some new tests managed to sneak in
usage again.
Bug: webrtc:10774
Change-Id: I354b4f5193625c8ddc75d54a252360810c3f60c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146983
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28697}
This is experimental field trial to exclude transport sequence number from FEC packets and should only be used in conjunction with datagram transport. Datagram transport removes transport sequence numbers from RTP packets and uses datagram feedback loop to re-generate RTCP feedback packets, but FEC contorol packets are calculated before sequence number is removed and as a result recovered packets will be corrupt unless we also remove transport sequence number during FEC calculations.
This change is a bit embarrassing, but it was the easiest workaround we found to make FEC work with datagrams. Added TODO to find better long term solution.
TODO(sukhanov): We need to find find better way to implement FEC with datagram transport, probably moving FEC to datagram integration layter. Wealso remove special field trial once we switch datagram path from RTCConfiguration flags to field trial and use the same field trial for FECworkaround.
Bug: webrtc:9719
Change-Id: I1e23c56e3cbaa087460410942fb6c5b4921a763e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146221
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28686}
These RTP header extensions are used for Unified Plan SDP / BUNDLE and
replace SSRC signaling.
Previously, the RTPSender would attach these header extensions to every
packet when configured. Now, the header extensions will be attached to
every packet until the an RTCP RR is received on that SSRC which
indicates the receiver knows what MID/RID the SSRC is associated with.
This should reduce overhead by 2-4 bytes per packet when the MID header
extension is used and by 4-8 bytes when both header extensions are used.
Bug: webrtc:10078
Change-Id: I5fa3ce28a75224adf11d2792bf4ff8dc76e46d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146480
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28685}
packet_size() includes the size of padding, this means that the size
check might incorrectly not trigger even if the payload is empty. In
turn this means that the ReadBigEndian call might read out of bounds
memory.
Refactored the code to reuse the App parsing code more, eliminating
the risk of this particular kind of error.
Bug: chromium:987507
Change-Id: Id8f3e292c3d30460d3cdb551f0a45070fdf8f022
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146716
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28680}
This interface is intended to only handle packet-sending parts of the
paced sender.
See https://webrtc-review.googlesource.com/c/src/+/145212 for context
Bug: webrtc:10809
Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28662}
This makes the code path where packets are directly owned by PacedSender
rather that being temporarily put in the RtpPacketHistory the default.
Functionally, this should essentially be a noop, with only minor timing
differences.
The old code-path will stay around for a short while and then be
removed once we are certain there are no regressions.
Bug: webrtc:10633
Change-Id: Id6360dea48fd0c9d46fde6f5eee93726d4f11d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146212
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28660}
If the SSRC of an RTP module is changed at runtime, we may get conflicts
with packets already there. Eg:
* Put seq# 123 in the history for SSRC 1.
* Change the SSRC to 2.
* Send a NACK for seq# 123 from SSRC 2.
Currently, we will respond with the packet belonging to SSRC 1 (and not
if the NACK specifies SSRC 1, to boot).
We can gen similar issues if the sequence number is changed, where
half frame are left in the buffer.
In these cases, the stream is likely being reset so we should just
clear the packet history too.
Bug: webrtc:10794
Change-Id: I28147c2532cf1c78840d4808c4366d4a647541f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145729
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28658}
This CL fixes two issues related to the TransmissionOffset header
extension and the new (not yet active) pacer mode.
Previously capture time (if unset) would be populated when put into the
packet history before entering the pacer. Since the pacer now owns the
packets, this does not occur until packet is actually sent, if at all.
Capture has really nothing to do with the packet history, this should
be set by the RtpSender pre-pacing instead.
Furthermore, for retransmissions the old path would take the capture
time from the original packet, build the RTX-wrapped retransmission and
set the toffset extension of the RTX packet using that captured capture
time. Since RTX packets are now fully built before the pacer, this does
not work, and we need to transfer the capture time from the original to
the RTX packet instead.
Bug: webrtc:10633
Change-Id: I031e8b6cc4ab20fb094dbd46720829b78951e7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146218
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28657}
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.
The functionality is negotiated using SDP.
It's added with a field trial that allow disabling the functionality in
case there's any issues.
Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
This CL removes the field trial left in place as a kill-switch in case
there were any regressions related to selecting payload padding based
on the likelihood of being useful instead of matching size.
It also removes the functionality that was only enabled with the
kill-switch active.
The feature has been default-on since June 23rd 2019:
https://webrtc.googlesource.com/src.git/+/214f54365ec210db76218a35ead66c9ce23e068e
Since we have not observed any issues, let's clean this code up.
Bug: webrtc:8975
Change-Id: I7f49fe354227b3f6566a250332e56b6d70fe2f09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145821
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28616}
This CL fixes two things related to the (not yet active) new
PacedSender code path:
1. Make sure BWE header extensions are properly populated for all
padding packets.
2. When generating padding, don't hold the RtpSender critsect when
accessing the RtpPacketHistory as this may lead to a lock order
inversion.
Bug: webrtc:10633
Change-Id: I8650fbf5dafddbeae61837d2137338163e1c48ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145723
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28613}
This reverts commit 11820502b8.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Make new pacer padding more like old one"
>
> This reverts commit bb7727211c.
>
> 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}
TBR=mbonadei@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Change-Id: I1fff79d75dc888921c6fbfc7f3980395a67e1c1a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145403
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28565}
This reverts commit 66147e892d.
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 6f129b3b76.
>
> 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 8b3e4e2d11.
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 6f420e4248.
>
> 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 94c58fd815
> >
> > 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 6f420e4248.
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 94c58fd815
>
> 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 6f129b3b76.
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 bb7727211c.
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 7325bc3917.
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 94c58fd815
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}
