This reverts commit 9e380fd484.
Reason for revert: breaking downstream projects
Original change's description:
> Improve performance of RtpPacketHistory
>
> The data structures in RtpPacketHistory were chosen based on assumption
> of few packets with possible sparse segments due to missing acking.
> In practice high bitrate usages with full histories seem to be more of
> a problem.
> Due to that, change storage from an std::map to an std::deque and live
> with potential segments of nullptr. Also limit size of padding prio
> set so that doesn't become a bottleneck.
>
> Bug: webrtc:8975
> Change-Id: I3b6314fb3255937d25362ff2cd906efb7b1397f7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145901
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29117}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I5d5b74a6f4d60588e01a52dafe33e26deb9bdf77
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152220
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29121}
The data structures in RtpPacketHistory were chosen based on assumption
of few packets with possible sparse segments due to missing acking.
In practice high bitrate usages with full histories seem to be more of
a problem.
Due to that, change storage from an std::map to an std::deque and live
with potential segments of nullptr. Also limit size of padding prio
set so that doesn't become a bottleneck.
Bug: webrtc:8975
Change-Id: I3b6314fb3255937d25362ff2cd906efb7b1397f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145901
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29117}
Reland with fixes for fuzzer found crashes.
This refactoring helps to reduce unnecessary memcpy calls on the receive side.
This CL replaces |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class, removes |length| field there, and does necessary changes.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
Bug: webrtc:10750
Change-Id: I6775a701bcb2ae25ec1666e1db90041cd49013b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151131
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29116}
Update the |cumulative_lost_| counter per received packet. The rules
follow from RFC 3550 and are fairly simple: Decrement the counter by
one for every received packet. For every in-order packet, i.e., increasing
|received_seq_max_|, add that change to |cumulative_lost_|.
Net change is zero as long as packets are received in proper sequence.
This way, GetStats() always returns an up-to-date value, independent
of the timing of RTCP report blocks.
For RTCP reports, keep a workaround to never report negative cumulative loss.
Bug: webrtc:10679
Change-Id: I47ff3bf266ff2382f405ec9828d34f7fad7068b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150641
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29058}
These fixes are automatically created by various analysis tools, but have been manually triggered to be applied.
* the 'empty' method should be used to check for emptiness instead of 'size' (3 times)
* using decl 'Return' is unused (4 times)
* using decl '_' is unused (3 times)
* using decl 'DoAll' is unused (2 times)
* using decl 'SetArgPointee' is unused
* using decl 'Dlrr' is unused
* using decl 'IsEmpty' is unused
* redundant get() call on smart pointer
* using decl 'Invoke' is unused (2 times)
* using decl 'SizeIs' is unused (3 times)
* using decl 'make_tuple' is unused
* using decl 'NiceMock' is unused
* using decl 'SaveArg' is unused (2 times)
* using decl 'AtLeast' is unused
* using decl 'ElementsAre' is unused
* using decl 'Gt' is unused
Bug: None
Change-Id: I97658fb0e94620b8319d7c3da29b15e27ec23188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151133
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29056}
This reverts commit eec5fff4df.
Reason for revert: Some crashes found by the fuzzer
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 replaces
> |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class,
> removes |length| field there, and does necessary changes.
>
> This is a reland of these two CLs with fixes:
> https://webrtc-review.googlesource.com/c/src/+/144942
> https://webrtc-review.googlesource.com/c/src/+/144881
>
> Bug: webrtc:10750
> Change-Id: I76f6dee5a57ade59942ea2822ca4737edfe6438b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29035}
TBR=brandtr@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org
Change-Id: Id3d65fb1324b9f1b0446fe217012115ecacf2b40
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151130
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29043}
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
This refactoring helps to reduce unnecessary memcpy calls on the receive
side.
This CL replaces
|uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class,
removes |length| field there, and does necessary changes.
This is a reland of these two CLs with fixes:
https://webrtc-review.googlesource.com/c/src/+/144942https://webrtc-review.googlesource.com/c/src/+/144881
Bug: webrtc:10750
Change-Id: I76f6dee5a57ade59942ea2822ca4737edfe6438b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29035}
This reverts commit 7db900e2e7.
Reason for revert: Speculative revert
Original change's description:
> Simplify pacer queue
>
> This CL simplifies the pacer queue by removing the now unnecessary
> beginpop/cancelpop/finalizepop methods. Instead there's a const top()
> and a pop() much like an stl queue.
> Old methods using the deprecated pacing code path are cleaned away.
>
> Bug: webrtc:10633
> Change-Id: Ib6da4d46a571bf56415172b790cc9e3f63206a38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150522
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28997}
TBR=sprang@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10633
Change-Id: I38f61afed4f4d542e236bcce3152a3aab52c6e6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29030}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
This CL simplifies the pacer queue by removing the now unnecessary
beginpop/cancelpop/finalizepop methods. Instead there's a const top()
and a pop() much like an stl queue.
Old methods using the deprecated pacing code path are cleaned away.
Bug: webrtc:10633
Change-Id: Ib6da4d46a571bf56415172b790cc9e3f63206a38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150522
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28997}
This interface/config field is now unused, let's remove it.
Bug: webrtc:10633
Change-Id: I56ff3d47ba784d973de411ada52ec9485bad9864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150531
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28978}
Number of received FEC bytes is used for the
WebRTC.Video.FecBitrateReceivedInKbps UMA histogram. Before this cl,
that value is based on a FEC packet counter updated by
ReceiveStatistics::FecPacketReceived. This cl deletes that method, and
instead adds a byte count to the FecPacketCounter struct, which is
maintained by the UlpFecReceiver and used for other FEC-related stats.
Bug: webrtc:10917
Change-Id: I24bd494b6909a2fe109d28e2b71ca8f413d05911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150533
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28976}
Previously the kDontRetransmit value was used to indicate packets that
should not be retransmitted but were put in the RtpPacketHistory anyway
as a temporary storage while waiting for a callback from PacedSender.
Since PacedSender now always owns the delayed packets directly, we can
remove all usage of StorageTye in RtpPacketHistory, and only put
packets there after pacing if RtpPacketToSend::allow_retransmission()
returns true.
Bug: webrtc:10633
Change-Id: I003b76ba43bd87658cc2a39e908fd28ebcd403f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150521
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28974}
This CL removes the old non-paced code path and instead uses a helper
class to just immediately pass the packet through the same code path as
when an actual pacer is used.
Bug: webrtc:10633
Change-Id: Id9a3ee4719829ad07710f5468e5452aa4bc8570b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150530
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28963}
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}
