Commit Graph

232 Commits

Author SHA1 Message Date
dd837e28fa Remove //rtc_base:timeutils from public deps
Bug: webrtc:8603
Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-06 11:23:21 +00:00
57aa81bce7 Remove //rtc_base:stringutils from public deps
Bug: webrtc:8603
Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-05 22:42:19 +00:00
e10a9f609a Remove //rtc_base:safe_conversions from public deps
Bug: webrtc:8603
Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-05 20:04:59 +00:00
33d31fbc48 Remove //rtc_base:rtc_event from public deps
Bug: webrtc:8603
Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-05 17:34:09 +00:00
4467ad7835 Remove //rtc_base:macromagic from public deps
Bug: webrtc:8603
Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-05 12:36:12 +00:00
0af55ba60d Remove //rtc_base:logging from public deps
Bug: webrtc:8603
Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-05 10:31:19 +00:00
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
f2b987377b in RtcpTransceiver implement sending rtcp sender reports
Bug: webrtc:8239
Change-Id: Id3298bf4e0eb18a3fc8072fb19416e67a126705f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249788
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35995}
2022-02-14 15:58:40 +00:00
47ff35db61 Delete some unneeded references to module.h and module_api target.
Bug: None
Change-Id: Iad983efa8666ca29b5e1ac22b3ade8987dbb2841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242371
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35648}
2022-01-10 13:40:24 +00:00
af1038d97c Allow providing the absolute capture time extension when packetizing a frame.
Bug: b/150859541
Change-Id: Iffb6ee84f49ffa64fdb0633248864d2dfd6e9ff3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234868
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35194}
2021-10-13 12:11:49 +00:00
68dc02b32f rtp_rtcp: use webrtc::flat_map for remote_senders
This one is frequently accessed - Mainly by ::CreateReportBlocks and
is visible in performance profiles (although not very much).

By using webrtc::flat_map, better data cache locality is expected.

Bug: webrtc:12689
Change-Id: Ic2ebcad806788074b2b4cb244a25395a48df1852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232541
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35054}
2021-09-21 13:20:16 +00:00
61a5bcbaa1 Migrate Dependency Descriptor parser from BitBuffer to BitstreamReader
BitstreamReader itself uses idea of Read function that always succeed,
and a separate function to check for errors.
Thus extra layer in the DependencyDescriptorReader is not needed.

Bug: None
Change-Id: Ie58861f2cbecc02a5a1a9538232494b4442c9afd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231226
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34940}
2021-09-07 14:19:17 +00:00
b8dce95006 Migrate VideoRtpDepacketizerVp9 implementation from BitBuffer to BitstreamReader
The new version is subjectivly cleaner
and objectively generates smaller binary size

Bug: None
Change-Id: I662596c41f30690f7e81a51804dd7feb263510c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231130
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34935}
2021-09-07 09:53:07 +00:00
9fdcfe90f1 In RtcpTransceiver add support for receiving network generic messages
These message suppose to extract all information
NetworkControllerInterface may need from rtcp.

Bug: webrtc:8239
Change-Id: I21d9081ad147ca8abe1ae05ca7201568c6ff77d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230421
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34876}
2021-08-30 13:58:18 +00:00
5ce7d14f81 Delete legacy rtp header parser as no longer used
Bug: None
Change-Id: I3c532eee7f2d9e5295874dd538730625c8d423ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227086
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34676}
2021-08-09 12:14:52 +00:00
18c0cc2bbd Refactor PacketSequencer in preparation for deferred sequencing.
This CL is extracted from
https://webrtc-review.googlesource.com/c/src/+/208584
PacketSequencer now has its own unit tests. They are maybe somewhat
redundant with a few RtpSender unit tests, but will defer cleanup to
a later CL.

Bug: webrtc:11340, webrtc:12470
Change-Id: I1c31004b85ae075ddc696bdf1100d2a5044d4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227343
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34638}
2021-08-03 20:00:39 +00:00
f715618eee Use flat_map in RTCPReceiver
RTCPReceiver initially used a std::map, which made
RTCPReceiver::IncomingPacket's use of std::map represent ~0.45% CPU in
highly loaded media servers. Using std::unordered_map in change 216321
reduced it only slightly, to 0.39%.

This is the second attempt to reduce it even further. By using a
flat_map and taking advantage of the increased cache locality, the hope
is that it will be reduced. These maps generally have low cardinality
(indexed by SSRC), and are looked up often, but modified less often,
which make them a potential candidate for flat_map.

Bug: webrtc:12689
Change-Id: I6733ccf3484d1c54e661250fb6712971b80fa2a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225203
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34432}
2021-07-07 13:43:59 +00:00
18649971ab Use flat_map in ReceiveStatisticsImpl
std::unordered_map represents ~0.57% CPU in a loaded media server,
which is expected to be reduced by using flat_map and its increased
cache locality compared to std::unordered_map, which use quite a few
allocations and indirections.

The number of SSRCs tracked by this class is expected to be low and
infrequently updated, but as GetOrCreateStatistician is called for every
incoming RTP packet, lookups are frequent.

Bug: webrtc:12689
Change-Id: I9a2c3798dcc7822f518e8f2624e78fceacd12d27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225202
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34430}
2021-07-07 08:34:45 +00:00
00ca0044d4 Unify helpers IsRtpPacket and IsRtcpPacket
Bug: None
Change-Id: Ibe942de433435d256cd6827440136936d4b274d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225022
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34419}
2021-07-06 10:39:00 +00:00
4e513346ec AV1 OBU test helper.
Bug: none
Change-Id: I942319122f823e18e500c049274527b00e6feba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223061
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34363}
2021-06-23 13:43:50 +00:00
6e65f6a428 Deprecating AbsoluteCaptureTimeReceiver
Bug: chromium:1056230, webrtc:10739
Change-Id: I42b6a6f1c61eaaa468898a09bb7add30f0a419fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223065
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34357}
2021-06-22 14:44:04 +00:00
fe3dd51f32 Use the injected clock in rtcp_transciever.
Bug: webrtc:11327
Change-Id: Idb02842f2eb679f972c0449a01a81a26ceb85827
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219789
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#34080}
2021-05-21 17:34:33 +00:00
63b3095d2b Make local to capturer clock offset a separate entry in PacketInfo.
This also changes the meaning of |estimated_capture_clock_offset| in
|absolute_capture_time_| to become a remote to capturer clock offset.

Bug: chromium:1056230, webrtc:10739
Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-20 13:42:57 +00:00
2497a27b22 Store RtpPacketReceived::arrival_time as Timestamp.
Previously this value was rounded up to a millisecond value.
This change is complementary to another change:
https://webrtc-review.googlesource.com/c/src/+/216398

Bug: webrtc:12722
Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33928}
2021-05-05 16:22:33 +00:00
dec73a8164 Add pure RtpSenderEgress unit test fixture.
The extracts and refactors some test code from rtp_sender_unittest.cc
and puts it in a new target intended to only test RtpSenderEgress, and
do it as pure unit test, rather than the unholy
not-quite-unit-not-quite-integration-test thingy we have today.

Only a first test case is actually ported with this CL, but it's a
start...

Bug: webrtc:11340
Change-Id: Ie2cdde63a00a6ff6eba7b8d443eeb76ce2a527c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216180
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33841}
2021-04-27 10:11:40 +00:00
bc1c93dc6e Add remote-outbound stats for audio streams
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats`
via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams.

`GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and
`ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`.

Bug: webrtc:12529
Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33452}
2021-03-12 20:39:50 +00:00
77ee8542dd Extract sequencing from RtpSender
This CL refactors RtpSender and extracts handling of sequence number
assignment and timestamping of padding packets in a separate helper
class.
This is in preparation for allowing deferred sequencing to after the
pacing stage.

Bug: webrtc:11340
Change-Id: I5f8c67f3bb90780b3bdd24afa6ae28dbe9d839a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208401
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33316}
2021-02-22 14:00:06 +00:00
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.

Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
2021-02-22 12:53:23 +00:00
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
e7c79fd3d6 Remove from chromium build targets that are not compatible with it.
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.

`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.

Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 13:46:19 +00:00
5312a8f532 Add option to attach custom object to an rtp packet
As an alternative to attaching custom array of bytes.

Bug: b/178094662
Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33069}
2021-01-25 18:31:34 +00:00
3e9cb2cbf2 Move deprecated code to their own build targets.
Moves the deprecated version of RtpRtcp module, and related classes
in video/.

Bug: webrtc:11581
Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33025}
2021-01-18 13:09:47 +00:00
098da17f35 Reland "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code"
This is a reland of 8c2250eddc7263036397179a0794b9b17d7afb38

Original change's description:
> Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
>
> Bug: webrtc:12336
> Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32978}

Bug: webrtc:12336
Change-Id: I1cd017d45c1578528dec4532345950e9823f4a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201732
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33003}
2021-01-15 17:59:05 +00:00
4319b1695e Revert "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code"
This reverts commit 8c2250eddc7263036397179a0794b9b17d7afb38.

Reason for revert: breaks downstream project

Original change's description:
> Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
>
> Bug: webrtc:12336
> Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32978}

TBR=danilchap@webrtc.org,mbonadei@webrtc.org,crodbro@webrtc.org

Change-Id: I5c9d419785254878a825865808b56841cd30b9b5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12336
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201731
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32979}
2021-01-14 15:02:47 +00:00
8c2250eddc Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
Bug: webrtc:12336
Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32978}
2021-01-14 14:32:26 +00:00
d73426d660 Add new empty build targets rtp_rtcp_legacy and video_legacy.
Initial step to be able to land
https://webrtc-review.googlesource.com/c/src/+/196904

Bug: webrtc:11581
Change-Id: Iaab52e98f4562f701cf02e3f641b7b02a11b799e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197944
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32971}
2021-01-14 09:45:26 +00:00
20e4c80fbe Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447

This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:29:36 +00:00
a94348440b VideoStreamEncoder report VideoLayersAllocation for simulcast
Adds support for Vp8 simulcast.

Bug: webrtc:12000
Change-Id: Ib24fd0542642b023ec35f7a7bdc4880d72365edf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187341
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32416}
2020-10-15 17:39:06 +00:00
0abd518abd Revert "Introduce RTC_NO_UNIQUE_ADDRESS."
This reverts commit f5e261aaf65cdf2eb903cdf40d651846be44f447.

Reason for revert: Breaks downstream projects.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
2020-10-07 07:37:01 +00:00
6556ed2402 Add experimental extension RtpVideoLayersAllocation
The extension is suggested to be used for signaling per target bitrate, resolution
and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting.
It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share.

Bug: webrtc:12000
Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32313}
2020-10-05 13:38:13 +00:00
f5e261aaf6 Introduce RTC_NO_UNIQUE_ADDRESS.
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.

The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.

Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
2020-09-30 09:52:49 +00:00
d381eede92 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.

Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
2020-09-07 08:37:14 +00:00
cccd55094d Delete unneeded dependencies on deprecated build target webrtc_common
Bug: webrtc:7660
Change-Id: Iad32aad8432fa2c6b3018d511b51943f869fbd11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182420
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31986}
2020-08-25 07:33:12 +00:00
2549f174b5 Remove RTPFragmentationHeader creation and propagation through webrtc
Bug: webrtc:6471
Change-Id: I5cb1e10088aaecb5981888082b87ae9957bbaaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31955}
2020-08-17 16:37:33 +00:00
e7c015e112 Migrate modules/rtp_rtcp to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31645}
2020-07-07 12:13:47 +00:00
62032d4592 Updates rtp_sender_unitests to use separate thread for pacer calls.
Some classes such as RtpSenderEgress makes assumptions about which
threads (e.g. paced sender vs worker thread) call specific methods.
Unit tests currently are single threaded so these checks are
essentially noops.

This change uses a separate task queue for calls epected to be called
by the pacer, so that inconsistencies in thread can be detected early.

Bug: None
Change-Id: Ic0904304a67eb034033524e62306da34b9eab8b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178602
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31628}
2020-07-03 19:51:58 +00:00
ba0ba71e93 Add 1 sec timer to ModuleRtpRtcpImpl2 instead of frequent polling.
This reduces the number of times we grab a few locks down from
somewhere upwards of around a thousand time a second to a few times.

* Update the RTT value on the worker thread and fire callbacks.
* Trigger NotifyTmmbrUpdated() calls from the worker.
* Update the tests to use a GlobalSimulatedTimeController.

Change-Id: Ib81582494066b9460ae0aa84271f32311f30fbce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177664
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31602}
2020-07-01 11:24:31 +00:00
096c0b0921 Post stats updates in RtpSenderEgress to the worker.
On the way remove need for lock for
rtp_sequence_number_map_ and timestamp_offset_.

Change-Id: I21a5cbf6208620435a1a16fff68c33c0cb84f51d
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177424
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31581}
2020-06-29 15:35:09 +00:00
473bbd8131 Remove a timer from ModuleRtpRtcpImpl2 that runs 100 times a second.
The timer fired a Notify call that goes to an object that already
receives callbacks for every packet from RtpSenderEgress.

Further optimizations will be realized by moving ownership
of the stats to the worker thread and then be able to remove
locking in a few classes that currently are tied to those
variables and the callbacks that previously did not come
from the same thread consistently.

We could furthermore get rid of one of these callback interfaces
and just use one.

Bug: webrtc:11581
Change-Id: I56ca5893c0153a87a4cbbe87d7741c39f9e66e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177422
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31575}
2020-06-29 08:09:14 +00:00
ee5b6de5aa Add helper for DependencyDescriptor rtp header extension
to decide when to set active_decode_target_bitmask field

Bug: webrtc:10342
Change-Id: I348d7467a72b45651455f4574fe8fda3c77ebbae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177400
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31555}
2020-06-24 10:59:40 +00:00