This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33
ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.
Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}
Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
Original change's description:
> Only include overhead if using send side bandwidth estimation.
>
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}
Bug: webrtc:11298
Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30390}
This reverts commit 8c79c6e1af354c526497082c79ccbe12af03a33e.
Reason for revert: Introduced a Bug that can happen if the include overhead state changes between pushing and poping a packet from the pacer packet queue.
Original change's description:
> Only include overhead if using send side bandwidth estimation.
>
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}
TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org
Change-Id: I0cacbc26408b7bec5bc3855a628e62407c081117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30383}
Corresponding mock class is deleted rather than updated,
since it appears unused.
Bug: webrtc:8422
Change-Id: If1c6c5ed73abff0d2545e8666c4bb8b63ee5b53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/13862
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29505}
The PacedSender is being reworked and will need an interface so we can
inject different implementations of it.
This CL introduces a new RtpPacketPacer interface inside the pacing
module. This interface handles the details of _how_ packets should be
paced, such as pacing rates/account for audio/max queue length etc.
The RtpPacketSender interface exposed from the rtp_rtcp module handles
only the actual sending of packets.
Some minor cleanups are included here.
Bug: webrtc:10809
Change-Id: I150b1a6262306d99e3f9d5f0b4afdb16a50e5ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145212
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28699}
