This can be used to determine whether the bitrate of a given spatial and temporal layer has been set in the allocation, even if the value it's set to is zero.
GetBitrate still returns 0 if the queried layer does not have the bitrate set.
Bug: webrtc:8479
Change-Id: I1d982e211da9b052fcccdbf588b67da1a4550c60
Reviewed-on: https://webrtc-review.googlesource.com/17440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20852}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=danilchap@webrtc.org
Bug: None
Change-Id: Ib4694d183f04d675f2ea66d39661fdffb2a984f1
Reviewed-on: https://webrtc-review.googlesource.com/23604
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20846}
Specifically, I'm moving
histogram_percentile_counter.h
mathutils.h
mod_ops.h
moving_max_counter.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I78a999984a27ef935be2d7c3136475d5f209adda
Reviewed-on: https://webrtc-review.googlesource.com/20870
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20832}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator
If Webrtc-ClockEstimation experiment is enabled, median filtering is
applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
out random errors introduced by incorrect RTCP SR reports and networking delays.
Bug: webrtc:8468
Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595
NOTRY=TRUE
NOPRESUBMIT=TRUE
Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595
Reviewed-on: https://webrtc-review.googlesource.com/23263
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20708}
If Webrtc-ClockEstimation experiment is enabled, median filtering is
applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
out random errors introduced by incorrect RTCP SR reports and networking
delays.
Bug: webrtc:8468
Change-Id: Iec6d094d2809d1efeb0b9483059167d9a03880da
Reviewed-on: https://webrtc-review.googlesource.com/22682
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20682}
Conditional visibility is complex to maintain and it is not well
supported by other build systems.
This CL removes it and falls back on the more relaxed visibility value
("*" in this case).
It is not a problem because the targets that are using conditional
visibility are all marked as "testonly" and this is probably enough to
keep the build graph clean.
Bug: None
Change-Id: I2d2b5ac9a02d08c2863950116db455976ee1459c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14902
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20658}
as an alternative to NextPacket function
to allow cleaner iterating over stacked rtcp packets.
Bug: webrtc:5565
Change-Id: I261afe2684e5fcb5fa3e7bcce272fbefeebd0b66
Reviewed-on: https://webrtc-review.googlesource.com/21360
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20615}
This is a reland of 47836b4ebb8a5c71695b5ec07bffd5ee4e3bc2ff
Internal tests are synced with the fix.
Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
>
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
>
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
>
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
Bug: none
Change-Id: I6087a8b20a926296b30432d69251670120b2a20c
Reviewed-on: https://webrtc-review.googlesource.com/20940
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20591}
This reverts commit 47836b4ebb8a5c71695b5ec07bffd5ee4e3bc2ff.
Reason for revert: This breaks internal tests, reverting to check if they recover.
Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
>
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
>
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
>
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,ssilkin@webrtc.org
Change-Id: I67d083cf769974d8df8bd5d70942af97db578db9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/20501
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20565}
spatial_idx is not present in RTP header if there is no temporal or
spatial layering. But the parser sets spatial_idx to 0 in this case.
When reflector repacketizes such packets it writes layering indices
into outgoing packets. When packets arrive to receiver it thinks that
it deals with multi layer stream and passes it through special path
in Vp9 reference frame finder which never outputs inter frames.
I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
when there is no layer indices in RTP header. Related unit tests have
been modified as well.
Bug: none
Change-Id: I14498cafb4e57797577dc873298c35b243479f88
Reviewed-on: https://webrtc-review.googlesource.com/17980
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20560}
There is a crash happening in this neighbourhood, so adding
CHECKs to tease it out explicitly.
BUG=webrtc:8481
Change-Id: I79a2ec8fd838f4a4735a04496e363b72975919ec
Reviewed-on: https://webrtc-review.googlesource.com/17361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20520}
The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.
Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}
Always use the packetization formely known as kEqualSize.
The RTPFragementation header is ignored, which is no change
in behaviour, since the caller previously always passed null.
Bug: webrtc:6471
Change-Id: Id9e2f985280c2ee8cc33fcf0e5c1fc3ee61c1aff
Reviewed-on: https://webrtc-review.googlesource.com/15222
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20459}
RtcpTransceiver name reserved for thread-safe version that planned to
be wrapper of the RtcpTransceiverImpl
BUG=webrtc:8239
Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f
Reviewed-on: https://webrtc-review.googlesource.com/7920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20414}
This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
Remove REMB accessor as used for debug checks only.
Merge SetRembData and SetRembStatus(true) eliminating
state 'remb can be send, but no data available yet'
Bug: None
Change-Id: I4c1c19435657e5cde02a17de90ec6de9f00b7daf
Reviewed-on: https://webrtc-review.googlesource.com/7983
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20240}
Remove redundant null pointer checks.
move header fields validation when they passed in rather when used.
Validate all used fields from the header.
Bug: webrtc:8335
Change-Id: I20b132c6fb8966e49a5414fe757f74d504f4a61a
Reviewed-on: https://webrtc-review.googlesource.com/7400
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20194}
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.
BUG=webrtc:8159
Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
The two timestamps, rtp time and corresponding system time, are always
used together, for audio/video sync. The new method reads both
timestamps, without releasing a lock in between. Ensures that the
caller gets values corresponding to the same packet.
Bug: webrtc:7135
Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14
Reviewed-on: https://webrtc-review.googlesource.com/4062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20120}
RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits:
1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew).
2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful.
BUG=webrtc:8111
Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb
Reviewed-on: https://webrtc-review.googlesource.com/1365
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20118}
This would allow us to limit the visibility of RtpPacketReceived and RtpPacketToSend, when we only want to allocate memory to save the RTP header, and not the metadata.
TBR=danilchap@webrtc.org
Bug: webrtc:8111
Change-Id: Ic9339189ccc2081d82bdc8def0fb39677458356f
Reviewed-on: https://webrtc-review.googlesource.com/5521
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20075}
Remove PayloadUnion's public member variables, so that the outside
world has to go through the accessors.
This is good code hygiene in general. For example, it makes it
possible to make the audio and video states Optional, so that exactly
one of them can be live at any one time.
BUG=webrtc:8159
Change-Id: Ie617b9038f961b329bd67b45478ff33d97148447
Reviewed-on: https://webrtc-review.googlesource.com/4428
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20064}
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>.
Split LogSessionAndReadBack into three functions and create class to share state between them.
Split VerifyRtpEvent into one incoming and one outgoing version.
Originally uploaded as https://codereview.webrtc.org/2997973002/
Bug: webrtc:8111
Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3
Reviewed-on: https://webrtc-review.googlesource.com/5020
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20063}
Stop using PayloadUnion's public member variables, since a future CL
will make them private.
BUG=webrtc:8159
Change-Id: Ia3dada56be7ef00ed80f3733209b18c178a36561
Reviewed-on: https://webrtc-review.googlesource.com/4380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20027}
I need to replace the audio part of PayloadUnion with SdpAudioFormat,
but that's a non-trivially-deletable class and those just don't work
well in unions, especially unions that don't have a discriminator that
says which member is currently active.
This CL converts the union to a class, adds a discriminator, and
provides accessor functions. CL #2 in the series will change all
outsiders to use the accessors instead of the public member variables
directly, and CL #3 will remove the public member variables. (It needs
to be done in separate steps like this because PayloadUnion is
unfortunately part of the API, and just changing it all in one go
would break users.)
BUG=webrtc:8159
Change-Id: I38c44bbb21a2d38600cff59bf37d8d47dfdbce21
Reviewed-on: https://webrtc-review.googlesource.com/4340
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20025}
