this change is intentionally noop.
Goal is to minimize change that would replace the
SingleThreadedTaskQueueForTest with a regular task queue.
Bug: webrtc:10933
Change-Id: I6da768941af048de3716af13e41b8f0f1ccd4cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157892
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29569}
This is a reland of 96f3de094566f32d842be6dd0906f1d13b8c8825
Downstream test is fixed, this is a pure reland.
TBR=danilchap@webrtc.org,srte@webrtc.org
Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}
Bug: webrtc:11036
Change-Id: I0731339dfd0781cc7f2f7ca78ac903539f25ff9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29514}
This reverts commit 96f3de094566f32d842be6dd0906f1d13b8c8825.
Reason for revert: Downstream test is borked.
Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}
TBR=danilchap@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I31330fd68ab809ff3951573791e9a79b81599958
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29511}
Since SSRCs of RTP modules are now set at construction time, we can
use just a simple unordered map from SSRC to module in packet router.
Bug: webrtc:11036
Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29510}
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 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}
Some of the tests are currently flaky because FEC is disabled if the
RTT is <200 ms, and the simulated network is configured to use 100 ms
for the send transport, but nothing is configured for the receive
transport. This CL configures the receive transport to 100 ms delay.
Bug: webrtc:10920
Change-Id: I79995693ba73683406fa9ced92a7918e6c05473f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154571
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29394}
actual task queue implementation for these tests is intentionally unchanged for now.
while at it, change return type of created transports to unique_ptr to note passing ownership.
Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
This class doesn't strictly follow rtc::TaskQueue semantics,
which makes it surprising and hard to use correctly.
Please use TaskQueueForTest instead.
This CL follows usual deprecation process:
1/ Rename.
% for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i
2/ Annotate old name for downstream users and accidental new uses.
Bug: webrtc:10933
Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29045}
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 reverts commit 8b3e4e2d1166464f6b309f4fc533a29607d2771f.
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 6f420e424885dab1d9f885365ea9abea5cc4a901.
>
> 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 94c58fd815f0c7c6429aa53a79621ea9ef39c770
> >
> > 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 6f420e424885dab1d9f885365ea9abea5cc4a901.
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 94c58fd815f0c7c6429aa53a79621ea9ef39c770
>
> 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 is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
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}
- Don't reset encoder if max/min bitrate changed.
- Removed min/max bitrate DCHECKs from encoder wrappers.
- Reset encoder if start_bitrate changed. Only do this if encoding
has not yet started.
- Updated ReconfigureBitratesSetsEncoderBitratesCorrectly test.
- Removed EncoderSetupPropagatesCommonEncoderConfigValues test since it
was a subset of ReconfigureBitratesSetsEncoderBitratesCorrectly.
Bug: webrtc:10773
Change-Id: Id9cbb2ea229232fd95967819e2a937b26948de9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144028
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28446}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
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 is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
to clearly signal passed ownership.
Drop support for accepting nullptr clock to avoid copying the Configuration structure.
Update all calls in webrtc to the new factory function
Bug: None
Change-Id: Ic5a78da8e59ba3988a757a9d9634fa31499ce0db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125901
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26994}
This prepares for future cleanup of how RtpTransportControllerSend is
used.
Bug: webrtc:10365
Change-Id: Idefc7e60f83819627c83b397949c8434d93491b3
Reviewed-on: https://webrtc-review.googlesource.com/c/124783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26923}
This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6.
Reason for revert: Failing tests fixed.
Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}
TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de.
Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}
TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.
For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.
Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
The overshoot detector uses a simple pacer model to determine an
estimate of how much the encoder is overusing the target bitrate.
This utilization factor can then be adjuster for when configuring the
actual target bitrate.
Spatial layers (simulcast streams) are adjusted separately.
Temporal layers are measured separately, but are combined into a single
utilization factor per spatial layer.
Bug: webrtc:10155
Change-Id: I8ea58dc6c4871e880553d7c22202f11cb2feb216
Reviewed-on: https://webrtc-review.googlesource.com/c/114886
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26573}
It doesn't do anything any more, so it should be removed.
Bug: webrtc:9586
Change-Id: I0b320b6ce4f480ff8cb59451db29bcc77b882b5f
Reviewed-on: https://webrtc-review.googlesource.com/c/120813
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26507}
