The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
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}
This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call.
Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes.
If there is custom factory present in dependencies it will be used, otherwise default factory will be used.
Intention behind this change is to have ability to have custom QoS with custom parameters.
Bug: webrtc:12778
Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34181}
This reduces the number of threads allocated per PeerConnection when
more than one PC is needed.
Bug: webrtc:11598
Change-Id: I3c1fd71705f90c4b4bbb1bc3f0f659c94016e69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31347}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I08ff36bd689fa7c3716c8e7017bd571cc9f09f35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127642
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27125}
