Commit graph

17 commits

Author SHA1 Message Date
Erik Språng
1cd7391c00 Turning off a stream should results in target bitrate 0 signal
Bug: webrtc:9734, b:116850043
Change-Id: Ia7b4a8ecf2099c3026c83b06febca833d428d0a2
Reviewed-on: https://webrtc-review.googlesource.com/c/103981
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25014}
2018-10-05 12:00:26 +00:00
Niels Möller
44b384d013 Delete support for VoIP metrics (RFC 3611 4.7)
Bug: None
Change-Id: I2f3cd622d3863fa88a9e1971894eced8eeb777e6
Reviewed-on: https://webrtc-review.googlesource.com/c/103805
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25007}
2018-10-05 10:07:57 +00:00
Erik Språng
8782a58b78 Send rtcp target bitrate immediately on new bitrate allocation structure
If layers have been enabled or disabled, send immediate instead of on
next available report.

Bug: webrtc:9823
Change-Id: Ifd774641d4b8c03a9efa8ad48ff5e88328ed2ba9
Reviewed-on: https://webrtc-review.googlesource.com/c/103802
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24997}
2018-10-04 15:47:36 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
Danil Chapovalov
d264df587f Replace rtc::Optional with absl::optional in modules/rtp_rtcp
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $dir -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: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-15 09:53:35 +00:00
Jonas Olsson
9633cff81a Remove "webrtc_rtp" traces.
They have been disabled by default for years, and should have been made redundant by the event logs.

Bug: webrtc:8982
Change-Id: I491923cbc93378d28f5166d24756b335619d9c12
Reviewed-on: https://webrtc-review.googlesource.com/82800
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23598}
2018-06-13 14:46:24 +00:00
Erik Språng
eeaa8f929c Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets
Bug: webrtc:9271
Change-Id: Ic7415830588bef9d87bab92943460207890dada6
Reviewed-on: https://webrtc-review.googlesource.com/76960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 11:22:56 +00:00
Erik Språng
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
Mirta Dvornicic
b1f063db32 Handle Receiver Reference Time Report from multiple receivers.
Bug: webrtc:9122
Change-Id: I996f02da26b11a4829fda740fdd452470daf4d24
Reviewed-on: https://webrtc-review.googlesource.com/66781
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22871}
2018-04-16 09:54:07 +00:00
Jiawei Ou
3587b8302a Make RTCP report interval configurable
Bug: webrtc:8789
Change-Id: I79c9132123c946b030ed79c647b4329e81d6e6ae
Reviewed-on: https://webrtc-review.googlesource.com/43201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21837}
2018-02-01 10:12:11 +00:00
Danil Chapovalov
1de4b62955 Change RtpRtcp::SetRemb signature to match RtcpTransceiver::SetRemb
in particular change bitrate type to int64_t to follow style guide.

With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter

Bug: webrtc:8239
Change-Id: I9ea265686d7cd2d709f0b42e8a983ebe1790a6ba
Reviewed-on: https://webrtc-review.googlesource.com/32302
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21250}
2017-12-13 14:40:01 +00:00
Danil Chapovalov
f74d641fc1 Simplify setting/unsetting REMB in RtcpSender
follow up of https://webrtc-review.googlesource.com/c/src/+/7983

Bug: None
Change-Id: I408c21408478d801a769e2e9d5f2eb9408430a4b
Reviewed-on: https://webrtc-review.googlesource.com/12520
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20359}
2017-10-19 14:07:31 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
niklase@google.com
5adc73aad3 git-svn-id: http://webrtc.googlecode.com/svn/trunk@166 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:46:41 +00:00
niklase@google.com
77ae29bc81 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:22:19 +00:00