7 Commits

Author SHA1 Message Date
Stefan Holmer
64be7fa7d8 Move FecController to RtpVideoSender.
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}
2018-10-05 14:39:01 +00:00
Artem Titov
3229d65fd0 Switch webrtc users from deprecated ctors.
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}
2018-08-17 13:54:51 +00:00
Niels Möller
7008287219 Delete struct webrtc::PacketTime.
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}
2018-08-07 10:07:15 +00:00
Danil Chapovalov
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
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}
2018-06-15 12:09:49 +00:00
Erik Språng
097085140e Reland: 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.

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}
2018-03-14 17:03:25 +00:00
Ilya Nikolaevskiy
16cba5c18d Revert "Add ability to emulate degraded network in Call via field trial"
This reverts commit 31a12c557dcd84a31f9c3f2d8858d9646c2a3135.

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}
2018-03-14 10:52:01 +00:00
Erik Språng
31a12c557d 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}
2018-03-14 10:22:50 +00:00