This is a reland of fa79081dca9faa8322943641352d9d2fd1b1b445
It crashed due to inability to handle small timestamps in probe
estimator. This was fixed by moving history window check to avoid
subtracting from the timestamp.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
Bug: webrtc:9510
Change-Id: I3bf91222068e4fbb6aa159bfeb7a73e00bb6a0d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143165
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28347}
This CL also improves test coverage and fixes an issue where the
(until now) unused code path for useful padding did not respect the
lower bound packet sizes.
Bug: webrtc:8975
Change-Id: I065745ca7ac9f7098d796c6a015cd96f052ee94f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142801
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28343}
This reverts commit 79890ef91faaf5d26a04083e97a3cb1218677ab0.
Reason for revert: the sync buffer was actually not counted when the buffer level filter was updated since the value was rounded down to the closest whole packet.
Original change's description:
> Remove sync buffer length from FilteredCurrentDelayMs.
>
> The sync buffer length is already added when the buffer level filter is updated.
>
> Bug: webrtc:10736
> Change-Id: Icbd411d4fd7b16f31b800142d1b6a8de79365d91
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140950
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28261}
TBR=minyue@webrtc.org,jakobi@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10736
Change-Id: Ibf4ce566484ff01421b186e03fe97fe633ba066d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143167
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28335}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
This reverts commit fa79081dca9faa8322943641352d9d2fd1b1b445.
Reason for revert: Breaks downstream project.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
TBR=terelius@webrtc.org,srte@webrtc.org
Change-Id: I562365fc5d1da68326d603338ccc6371114d7e12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143164
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28331}
As the send time congestion controller now has been removed,
we don't need the RTP related constructs anymore.
Bug: webrtc:9510
Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28330}
instead of using factories for MediaEngine and RtcEventLog that rely on GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ie1135f70f4ae4d047c4d6bf2db61489a663385aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141875
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28328}
Users that would like custom TaskQueueFactory (e.g. chromium) should use
CreateModularPeerConnectionFactory directly
Bug: webrtc:10284
Change-Id: I7ba55a0f21560d4beb71a7f93f6fa70b0fe74931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142234
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28324}
This change is breaks the dependency between "api:rtp_headers" and "api/video:video_frame".
Bug: webrtc:10668
Change-Id: Ib827de2c0e33809ab1d58338037563aa2fa249bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140949
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28322}
This CL replaces all uses of Timestamp::us(Clock::TimeInMicroseconds())
with Clock::CurrentTime() which should be a no-op apart from slight
changes in checks.
Additionally instances of Timestamp::ms(Clock::TimeInMilliseconds()) in
test code is replaced. This slightly changes the behavior since the
timestamp will get increased resolution.
Timestamp::ms(Clock::TimeInMilliseconds()) in non-test code is untouched
to avoid changing behavior of production code.
Bug: webrtc:9883
Change-Id: I8047f4cb2ca735f44f11d32f9367aa3eb376ec04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142803
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28321}
outgoing checks.
This change adds an experimental feature to allow an ICE agent to embed
the transaction ID of the latest connectivity check received from the
remote peer, as an auxiliary acknowledgement in additional to the check
response, in its own checks. This could facilitate the establishment of
ICE connectivity if the check process has a high RTT.
Bug: None
Change-Id: If3e6327720f13beeb14f103af3b5ffb4f9692998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142682
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28316}
In https://chromium-review.googlesource.com/1650265 attributes like minSdkVersion were moved from AndroidManifest.xml to GN files. For WebRTC there were a few problems with that.
* We don't want to suppress UsesMinSdkAttributes lint but now there are these "invalid" manifest files that we can't exclude or discern. So disable this lint error.
https://chromium-review.googlesource.com/c/chromium/src/+/1650265/14/build/android/AndroidManifest.xml
* We should specify the versions in GN files, so I did that here (by exactly copying the versions that are already in the targets' corresponding XML files), but we never want to get rid of them in the XML files. For now this information will just be duplicated (without any synchronicity check!) so there should be followup to this.
Change log: 6ae0f0cd4c..bf62d746a4
Full diff: 6ae0f0cd4c..bf62d746a4
Changed dependencies
* src/base: 9e5e9332df..e5a1d1f652
* src/build: 5a031748ec..2ef566e990
* src/buildtools: 6ae683be2f..6f3775ad6e
* src/buildtools/linux64: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/mac: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/win: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/ios: 2f5c817266..7f1a97d593
* src/testing: 1d4247de57..b1b36ff0d4
* src/third_party: 6f7cbf7c46..42e96c4074
* src/third_party/android_sdk/public: ki7EDQRAiZAUYlnTWR1XmI6cJTk65fJ-DNZUU1zrtS8C..xhyuoquVvBTcJelgRjMKZeoBVSQRjB7pLVJPt5C9saIC
* src/third_party/android_sdk/public: iIwhhDox5E-mHgwUhCz8JACWQCpUjdqt5KTY9VLugKQC..ppQ4TnqDvBHQ3lXx5KPq97egzF5X2FFyOrVHkGmiTMQC
* src/third_party/android_sdk/public: 4Y2Cb2LGzoc-qt-oIUIlhySotJaKeE3ELFedSVe6Uk8C..MSnxgXN7IurL-MQs1RrTkSFSb8Xd1UtZjLArI8Ty1FgC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ed9fcf3f70..9e5dbd8b46
* src/tools: f58f33bca1..a9a4b8fc7b
DEPS diff: 6ae0f0cd4c..bf62d746a4/DEPS
No update to Clang.
Bug: chromium:891996
Change-Id: I773d6fa90e8083d934c84eecc1cb9d7d4496eca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142235
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28311}
This methods multiplies an int with 500. This cl ensure that does not overflow.
BUG=none
Change-Id: I2c4c4c169baf3bbc2cac021e87e396c605cd1815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141860
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28310}
This CL builds on https://webrtc-review.googlesource.com/c/src/+/142165
It adds the parts within the paced sender that uses those send methods.
A follow-up will add the pre-pacer RTP sender parts. That CL will also
add proper integration testing. Here, I mostly add coverage for the new
send methods. When the old code-path is removed, all tests need to be
converted to exclusively use the owned path.
Bug: webrtc:10633
Change-Id: I870d9a2285f07a7b7b0ef6758aa310808f210f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142179
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28308}
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.
Bug: webrtc:10749
Change-Id: I73e9402136cc16902d177234d63369938f191e5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142223
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28302}
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.
Bug: webrtc:10749
Change-Id: I4d8ec92643f1a0d4b5fdac2fbcb067766afc6548
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142222
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28300}
This method will be called when PacedSender is using the new code path
that directly owns the packets to be sent.
It can be seen as combining a few features of the old code path:
* It checks if this is the correct RTP module and then sends, without
the need for PacketRouter to poll multiple methods for SSRC etc first.
* It partly corresponds to TimeToSendPacket(), but RTX encapsulation
now happens pre-pacer and FEC does not need to have a packet history,
so most of that method is not used.
* It implements most of PrepareAndSendPacket(), such as updating header
extensions, reporting stats and of course forwards to Transport. It
now also handles the history a bit differently, since media packets
will only be stored for potential retransmission post-pacer.
Bug: webrtc:10633
Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28298}
