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}
We should reject invalid values explicitly in order to prevent DCHECK
failures later, which affect fuzzing progress.
Bug: chromium:1009172, chromium:1009073
Change-Id: I7f0dc417ecac7aab076a652143f5face2ff98da2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156340
Commit-Queue: Kuang-che Wu <kcwu@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29459}
use frame_type from the RTPVideoHeader instead of as an extra parameter
merge payload data and payload size into single argument
pass RTPVideoHeader by value (relying on copy elision)
Bug: None
Change-Id: Ie7970af3b198b83b723d84c7a8b047219c4b38c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29445}
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}
This will allow RtcpPackets to be sent in a more generic way where the
PacketRouter does not have to know about the type.
App::SetSsrc is replaced with SetSenderSsrc
Bug: webrtc:10742
Change-Id: I9fa18d408250f15818dc6898093d9b116603facb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156166
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29420}
Add missing includes to files that were transactivly depending on removed includes.
Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
This CL moves the various parameters in the the RTPSenderVideo ctor into
a struct, and adds the red/ulpfec payload types to it.
Once the downstream usage of SetUlpfecConfig() is gone, we can make
those members const and avoid locking in SendVideo().
Bug: webrtc:10809
Change-Id: I9a96ab5b2a4eb2997ebf4a3a3e3cd2eb5715fd79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155365
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29384}
This CL removes some unnecessary locking, since we are already
serialized by the lock in VideoStreamEncoder. A simple RaceChecker is
used to verify this.
We also remove the usage of RegisterPayloadType() and replace it with
a parameter in SendVideo instead. This way we are prepared for removing
the payload type map and lock entirely. Some usage still exists
downstream and needs to be removed before cleaning this up.
Bug: webrtc:10809
Change-Id: Ie90163f15d11c8843f3beaf9a0df0dd2a1fd5ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154700
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29372}
This is a reland of a21d50c1f3eab29fd9026cc67c8cb4017efda5e3
Original change's description:
> Prepares RtpSenderVideo for batch forwarding of generated packets
>
> In order to reduce contention, this CL avoids taking locks per packet
> and prepares for forwarding all packets for a frame in one call, rather
> than one at a time. This will especially reduce contention in the paced
> sender during very high packet rates.
>
> Bug: webrtc:10809
> Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29323}
Bug: webrtc:10809
Change-Id: I50e0a27eb3b0b1afa39f250febdd564e1e1f06eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155362
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29367}
This reverts commit a21d50c1f3eab29fd9026cc67c8cb4017efda5e3.
Reason for revert: Speculative revert due to unexpected perf changes.
Original change's description:
> Prepares RtpSenderVideo for batch forwarding of generated packets
>
> In order to reduce contention, this CL avoids taking locks per packet
> and prepares for forwarding all packets for a frame in one call, rather
> than one at a time. This will especially reduce contention in the paced
> sender during very high packet rates.
>
> Bug: webrtc:10809
> Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29323}
TBR=ilnik@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10809
Change-Id: I1cbf0ce0cc06f9195b5e0716b8dd4c85f7f6bab1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155164
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29341}
If the packet can't be parsed, the buffer isn't moved to the packet.
Then, a new empty buffer is moved back from the packet.
Thus, the consequtive DCHECK fails because the data isn't the same anymore.
Bug: chromium:1009236
Change-Id: Ie27f438c40f38074d42d8491fe03df45d50eba50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155162
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29340}
In order to reduce contention, this CL avoids taking locks per packet
and prepares for forwarding all packets for a frame in one call, rather
than one at a time. This will especially reduce contention in the paced
sender during very high packet rates.
Bug: webrtc:10809
Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29323}
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
"s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"
Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29282}
This is a reland of 9e380fd484db09c37323b90a19c5ce7965927975
Patchset 1 is the original CL. The follow-ups adds fix for a test failure
and test for that change.
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}
Bug: webrtc:8975
Change-Id: I5038e5ad2eb79ce75710d2d8b0b3ac01dd41c013
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152282
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29152}
This reverts commit 9e380fd484db09c37323b90a19c5ce7965927975.
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 eec5fff4df92b2330e5fec67ff08c7cbb4c4ab8d.
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 7db900e2e78d1644a173a0bc505ad52c61c43f9b.
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}
