This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
Downstream test now fixed.
As a precaution, also avoid DCHECKS for non-zero SSRC.
First patch set is reland, second makes checks more lenient.
Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}
Bug: webrtc:10774
Change-Id: I540b49a31a31e98d87f02ae04083d5206e71c1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157100
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29491}
This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.
Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.
This prepares for reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29490}
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
Downstream fixed, relanding.
Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}
TBR=nisse@webrtc.org
Bug: webrtc:10774
Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29486}
Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
remove them, make the members const, and remove now unnecessary locking.
Bug: webrtc:10774
Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29475}
This method sends arbitrary number rtp::RcpPackets into one or more IP packets.
It is implemented both in RtcpTranceiver and in RtpRtcp.
Change-Id: I00424ee2f1730ff98626f768846f4ac1ad864933
BUG: webrtc:10742
Change-Id: I00424ee2f1730ff98626f768846f4ac1ad864933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156240
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29430}
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}
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}
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}
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}
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}
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}
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}
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 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 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}
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}
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 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}
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 builds on https://webrtc-review.googlesource.com/c/src/+/142165
It adds the parts within the paced sender that uses those send methods.
A follow-up will add the pre-pacer RTP sender parts. That CL will also
add proper integration testing. Here, I mostly add coverage for the new
send methods. When the old code-path is removed, all tests need to be
converted to exclusively use the owned path.
Bug: webrtc:10633
Change-Id: I870d9a2285f07a7b7b0ef6758aa310808f210f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142179
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28308}
