This CL adds the ability to configure RTPSender to include the
MID header extension when sending packets. The MID will be
included on every packet at the start of the stream until an RTCP
acknoledgment is received for that SSRC at which point it will
stop being included. The MID will be included on regular RTP
streams as well as RTX streams.
Bug: webrtc:4050
Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f
Reviewed-on: https://webrtc-review.googlesource.com/60582
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22574}
This is a reland of 6328d7cbbc
Original change's description:
> Rework rtp packet history
>
> This CL rewrites the history from the ground up, but keeps the logic
> (mostly) intact. It does however lay the groundwork for adding a new
> mode where TransportFeedback messages can be used to remove packets
> from the history as we know the remote end has received them.
>
> This should both reduce memory usage and make the payload based padding
> a little more likely to be useful.
>
> My tests show a reduction of ca 500-800kB reduction in memory usage per
> rtp module. So with simulcast and/or fec this will increase. Lossy
> links and long RTT will use more memory.
>
> I've also slightly update the interface to make usage with/without
> pacer less unintuitive, and avoid making a copy of the entire RTP
> packet just to find the ssrc and sequence number to put into the pacer.
>
> The more aggressive culling is not enabled by default. I will
> wire that up in a follow-up CL, as there's some interface refactoring
> required.
>
> Bug: webrtc:8975
> Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> Reviewed-on: https://webrtc-review.googlesource.com/59441
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22347}
Bug: webrtc:8975
Change-Id: I162cb9a1eccddf567bdda7285f8296dc2f005503
Reviewed-on: https://webrtc-review.googlesource.com/60900
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#22356}
Reviewed-on: https://webrtc-review.googlesource.com/61661
Cr-Commit-Position: refs/heads/master@{#22438}
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}
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}
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}
Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d
