Old target and call/simulated.h exist but refer to new target in test/network.
Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
This reverts commit 897ea04db5.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
and in DegradedCall. In DegradedCall - ThreadPacketReceiver is no longer needed.
Implementation of DeliverRtpPacket is done in preparation of https://webrtc-review.googlesource.com/c/src/+/290540, where the parsed packet will be propagated to Call without extra parsing.
Bug: webrtc:7135, webrtc:14795
Change-Id: Ic068105d6d1f337afc6b4539b0e7184e736e7ee0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290704
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39048}
DeliverRtpPacket use a parsed RTP packet as argument where the RTP extensions are supposed to be known.
This method is implemented in webrt::Call and temporary used by the exising method Call::DeliverRtp, but the idea is to instead avoid extra packet parsing by forwarding a RtpPacketReceived from RtpTransport::DemuxRtpPacket via WebrtcVideoChannel::OnPacketReceived and WebrtcVoiceChannel.
DeliverRtcpPacket is also implemented in Call and is directly used in PeerConnection::InitializeRtcpCallback.
Bug: webrtc:14795, webrtc:7135
Change-Id: Ib6ffe8e1229ac07fa459ee2fc9a0af8455a23bac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290401
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39015}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
Ability to emulate degraded networks using DegradedCall has not
been covered by tests and it is crashing due to DCHECKs.
Fix threading issues so this no longer crash.
Bug: None
Change-Id: I9276dfb1f71762faa02146af0bfaab713bebb7f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276060
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38216}
This reverts commit 16a8b25d80.
Reason for revert: Checking if this is blocking the Chromium autoroller.
Original change's description:
> Update local_ssrc without needing to recreate video streams.
>
> This is comparable to this change done previously for for audio streams:
> https://webrtc-review.googlesource.com/c/src/+/222042
>
> Bug: webrtc:11993
> Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36876}
Bug: webrtc:11993
Change-Id: I3a8d2f6a7e89b6784754d8e891a4e01479807c2d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262422
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36892}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
This avoids creating a dedicated task queue that potentially contends
with the network thread.
Bug: webrtc:251043
Change-Id: I2d54486c4235dac44b79f89d4f3d81b4d3de4026
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252282
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36058}
The fake network configs are now specified using just two field trials:
WebRTC-FakeNetworkSendConfig and WebRTC-FakeNetworkReceiveConfig.
Both of them have the following parameters from
BuiltInNetworkBehaviorConfig:
* queue_length_packets // Queue length in number of packets.
* queue_delay_ms // Delay in addition to capacity induced delay.
* delay_standard_deviation_ms // Standard deviation of the extra delay.
* link_capacity_kbps // Link capacity in kbps.
* loss_percent // Random packet loss.
* allow_reordering // If packets are allowed to be reordered.
* avg_burst_loss_length // The average length of a burst of lost packets.
* packet_overhead // Additional bytes to add to packet size.
* codel_active_queue_management // Enable CoDel active queue management.
Plus:
* duration // For how long to use this config before progressing.
Example:
WebRTC-FakeNetworkSendConfig/queue_delay_ms:66|1,loss_percent:1|0,link_capacity_kbps:200|10000,queue_length_packets:10|100,duration:15s|45s/
This creates two configs:
1. For 15s, apply 66ms delay, 1% loss, 200kbps bandwidth, 10 packet queue size
2. For 45s, apply 1ms delay, 0% loss, 10Mbps bandwidth, 100 packets queue size
(then repeat)
Bug: webrtc:13655
Change-Id: I0524f572de480731df4d414724203772182c628b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251043
Reviewed-by: Stefan Holmer <holmer@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35952}
Classes associated with the Call instance, need access to these threads
and/or awareness, for checking for thread correctness.
Bug: webrtc:11993
Change-Id: I93bcee0657875f211be2ec959b96f818fa9fd8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215584
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33772}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
This CL adds AddAdaptationResource to Call and
AddAdaptationResource/GetAdaptationResources method to relevant
VideoSendStream and VideoStreamEncoder interfaces and implementations.
Unittests are added to ensure that resources can be added to the Call
both before and after the creation of a VideoSendStream and that the
resources always gets added to the streams.
In a follow-up CL, we will continue to plumb the resources all the way
to PeerConnectionInterface, and an integration test will then be added
to ensure that injected resources are capable of triggering adaptation.
Bug: webrtc:11525
Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31499}
Previously this functionality only worked correctly with a single
Transport instance, meaning a single video track.
This CL moves the transport pointer from being a member in
FakeNetworkPipe to being set on each packet, so that e.g. audio packets
point to the audio transport and video packet to the video transport.
This means we need a separate adapter per stream in DegradedCall.
Additionally, since Transport instances can potentially be destroyed
before it's time to forward the message to it, we need to keep track
of which instance that are live and ignore packets we can't forward.
Bug: None
Change-Id: I314d431c04ff81c3859cf661e2722c99342f785e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148586
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28831}
There aren't any tests for this and the code isn't currently
active except for the fact that it adds complexity to the Call
class, synchronization into the active code path and makes future
improvements to the class more complex or impossible.
Bug: webrtc:9719
Change-Id: Ia41af0b2186b8a36ca70a07858990b6af7f3a5c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148078
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28807}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
(so far SetBitrate did not do anything for media transport)
Bug: webrtc:9719
Change-Id: I48e669341ffe6c9e4697ff9146c314be7796a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27169}
Currently there's an implicit requirement that users of
SimulatedNetwork should call it repeatedly, even if the return value
of NextDeliveryTimeUs is unset.
With this change, it will indicate that there might be a delivery in
5 ms at any time there are packets in queue. Which results in unchanged
behavior compared to current usage but allows new users to expect
robust behavior.
Bug: webrtc:9510
Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069
Reviewed-on: https://webrtc-review.googlesource.com/c/120402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26617}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
Add TargetRateObservers for media transport in the call object.
Bug: webrtc:9719
Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7
Reviewed-on: https://webrtc-review.googlesource.com/c/110622
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25662}
This also moves the packet feedback tracking to RtpVideoSender.
Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
Stop using of deprecated ctors of FakeNetworkPipe in most part of
webrtc codebase, except DirectTransport, where further refactoring will
be continued in future CLs.
Bug: webrtc:9630
Change-Id: I823404469e461601ddbc026eaeac668eeda8045f
Reviewed-on: https://webrtc-review.googlesource.com/94763
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24327}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
TBR=stefan@webrtc.org, philipel@webrtc.org
Originally reviewed on: https://webrtc-review.googlesource.com/33013
Bug: webrtc:8910
Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0
Reviewed-on: https://webrtc-review.googlesource.com/61782
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22430}
This reverts commit 31a12c557d.
Reason for revert: Breaks downstream project.
Original change's description:
> Add ability to emulate degraded network in Call via field trial
>
> This is especially useful in Chrome, allowing use to emulate network
> conditions in incoming or outgoing media without the need for platform
> specific tools or hacks. It also doesn't interfere with the rest of the
> network traffic.
>
> Also includes some refactorings.
>
> Bug: webrtc:8910
> Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
> Reviewed-on: https://webrtc-review.googlesource.com/33013
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22418}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I22bda6da01c2ff5abd6f408c5ee9e4fba21294f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8910
Reviewed-on: https://webrtc-review.googlesource.com/61700
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22419}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
Bug: webrtc:8910
Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
Reviewed-on: https://webrtc-review.googlesource.com/33013
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22418}
