Commit Graph

57 Commits

Author SHA1 Message Date
9416ef8c4f Rename PayloadRouter to RtpVideoSender.
Bug: webrtc:9517
Change-Id: I18397a28067dbe5029fc80fe2eef360869abb339
Reviewed-on: https://webrtc-review.googlesource.com/89380
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24039}
2018-07-19 08:50:50 +00:00
dbdb3a0079 Refactoring PayloadRouter.
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
  VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
  of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
  renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.

Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
2018-07-17 14:46:15 +00:00
a2f1533e27 Moved PayloadRouter to call/.
This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend.

Bug: webrtc:9517
Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7
Reviewed-on: https://webrtc-review.googlesource.com/88241
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23936}
2018-07-11 15:38:39 +00:00
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
566124a6df Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.

Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.

Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-23 15:31:27 +00:00
652dc915bc Adds unit tests for VideoSendStreamImpl.
Bug: None
Change-Id: Ifadad47af4769d8aca42c98832cea49a6c7977cd
Reviewed-on: https://webrtc-review.googlesource.com/71040
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22959}
2018-04-20 13:12:13 +00:00
8e0b15b584 Moves VideoSendStreamImpl to a separate file.
This prepares for adding unit tests for VideoSendStreamImpl.

Bug: None
Change-Id: I488041b09f4a455ce4cf1bdc7b8163ef6ad19a8a
Reviewed-on: https://webrtc-review.googlesource.com/70782
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22926}
2018-04-18 20:44:36 +00:00