Commit graph

11 commits

Author SHA1 Message Date
Tommi
6be3e788f5 Add getter for rtp header extensions for receiver classes.
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.

Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
2022-05-09 16:59:19 +00:00
Tommi
cb7c7366d0 Separate reading remote_ssrc from using the rtp_config() getter.
`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.

Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
2022-05-09 14:55:00 +00:00
Tommi
1f38a38b6f Add ability to set rtp header extensions without recreating streams.
Setting the rtp header extensions on the packet delivery thread
(currently worker, soon to be network), is now possible without
taking the hit of deleting and recreating the receive stream (and
rtp receiver and related state).

Bug: webrtc:11993
Change-Id: I9bbe306844a25d85d79cd216092ead66eaf68960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223741
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34953}
2021-09-08 13:39:36 +00:00
Markus Handell
eb61b7f620 ModuleRtcRtcpImpl2: remove Module inheritance.
This change achieves an Idle Wakeup savings of 200 Hz.

ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.

Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
2021-06-22 14:51:04 +00:00
Tommi
d350006b70 Add rtp_config() accessor to ReceiveStream.
This is a consistent way to get to common config parameters for
all receive streams and avoids storing a copy of the extension
headers inside of Call. This is needed to get rid of the need of
keeping config and copies in sync, which currently is part of why
we repeatedly delete and recreate audio receive streams on config
changes.

Bug: webrtc:11993
Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34285}
2021-06-14 17:57:57 +00:00
Tommi
90738ddb4e Split VideoReceiveStream2 init into worker / network steps.
This is in preparation for actually doing this initialization
differently in the Call class. This CL takes the registration
steps that are inherently network thread associated and makes
them separate from the ctor/dtor.

Inject Call* instead of worker_thread(), which will simplify upcoming
work that needs to access the network_thread() as well.

This is related to:
https://webrtc-review.googlesource.com/c/src/+/220608
https://webrtc-review.googlesource.com/c/src/+/220609

Bug: webrtc:11993
Change-Id: I72769fd61de84967d9a645750c40d01660a2716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220764
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34172}
2021-05-31 17:10:23 +00:00
Tommi
0377bab21b Split FlexfecReceiveStreamImpl init into worker / network steps.
This is comparable to this change for AudioReceiveStream:
https://webrtc-review.googlesource.com/c/src/+/220608/

Bug: webrtc:11993
Change-Id: I6bad7fa693441f80e86d8b021b8cf42727dc9142
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220609
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34170}
2021-05-31 15:29:41 +00:00
Tomas Gunnarsson
f25761d798 Remove dependency from RtpRtcp on the Module interface.
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.

Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.

The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.

Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-04 08:11:21 +00:00
Sebastian Jansson
8026d60ea9 Injecting Clock in video receive.
Bug: webrtc:10365
Change-Id: Id20fca5b8ad13c133e05efa8972d8f5679507064
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125192
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26958}
2019-03-04 21:53:57 +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
Renamed from webrtc/call/flexfec_receive_stream_impl.h (Browse further)