This information is now readily available. Let's expose it.
In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.
Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
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}
The 'worker' noun in WebRTC is tied to the worker thread.
Hence naming an unrelated queue to something with worker
confuses code reading.
Change this to something which can't reasonably be confused
with the worker thread.
Bug: webrtc:11993
Change-Id: Icdcc728cf3dd9eb020f922367eebd0c520814568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220934
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34183}
This is a reland of 51e5c4b0f47926e2586d809e47dc60fe4812b782
It may happen that user will pass config with min bitrate > max bitrate.
In such case we can't generate cached_constraints and will crash before.
The reland will handle this situation gracefully.
Original change's description:
> Fix data race for config_ in AudioSendStream
>
> config_ was written and read on different threads without sync. This CL
> moves config access on worker_thread_ with all other required fields.
> It keeps only bitrate allocator accessed from worker_queue_, because
> it is used from it in other classes and supposed to be single threaded.
>
> Bug: None
> Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33125}
Bug: None
Change-Id: I274ff15208d69c25fb25a0f1dd0a0e37b72480b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33162}
This reverts commit 51e5c4b0f47926e2586d809e47dc60fe4812b782.
Reason for revert: Speculatively reverting because WebRTC fails to
roll due to a DCHECK in audio_send_stream.cc in a web platform test
and this is the only CL on the blamelist that touches that file.
Original change's description:
> Fix data race for config_ in AudioSendStream
>
> config_ was written and read on different threads without sync. This CL
> moves config access on worker_thread_ with all other required fields.
> It keeps only bitrate allocator accessed from worker_queue_, because
> it is used from it in other classes and supposed to be single threaded.
>
> Bug: None
> Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33125}
TBR=danilchap@webrtc.org,peah@webrtc.org,nisse@webrtc.org,hta@webrtc.org,titovartem@webrtc.org
# Initially not skipping CQ checks because original CL landed > 1 day
# ago. Adding NOTRY now because of ios_sim_x64_dbg_ios12 issues.
NOTRY=True
Bug: None
Change-Id: I33355198fca96faad7ac77538c7bd31425f46ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205340
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33142}
config_ was written and read on different threads without sync. This CL
moves config access on worker_thread_ with all other required fields.
It keeps only bitrate allocator accessed from worker_queue_, because
it is used from it in other classes and supposed to be single threaded.
Bug: None
Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33125}
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.
Transport-cc extension still needs to be negotiated.
Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
Potential deadlock fixed by acquiring lock before calling encoder.
This is a reland of a135557b3c7ffa4fb1710d2d907c3cb86c5d5913
Original change's description:
> Call OnReceivedOverhead after audio network adaptor is created.
>
> This prevents ending up in a state where audio network adaptor never
> receives the current packet overhead and therefore doesn't work.
>
> Bug: chromium:1086942
> Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31951}
Bug: chromium:1086942
Change-Id: I514e523c6607cee0099b87919f0f77ebec966ddd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181888
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31971}
This reverts commit a135557b3c7ffa4fb1710d2d907c3cb86c5d5913.
Reason for revert: Suspected downstream breakage
Original change's description:
> Call OnReceivedOverhead after audio network adaptor is created.
>
> This prevents ending up in a state where audio network adaptor never
> receives the current packet overhead and therefore doesn't work.
>
> Bug: chromium:1086942
> Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31951}
TBR=peah@webrtc.org,sprang@webrtc.org,jakobi@webrtc.org
Change-Id: I96a92f82f0431457d649cc7feb253f0e026eeada
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1086942
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181885
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31954}
This prevents ending up in a state where audio network adaptor never
receives the current packet overhead and therefore doesn't work.
Bug: chromium:1086942
Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31951}
The lock in TransportFeedbackProxy could cause a dead-lock if audio is
included in transport feedback messages, and necessitated a revert:
https://webrtc-review.googlesource.com/c/src/+/178100
This CL removes that lock and in fact the entire TransportFeedbackProxy
class, and instead sets the observer at construction time.
We therefore don't need to guard the observer pointer anymore.
For further context, see also internal bug b/153893626
Bug: webrtc:10809
Change-Id: I79b08d8d0e587a59736b383c3596a26836c33d2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178207
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31583}
negotiates the RED codec for opus audio behind a field trial
WebRTC-Audio-Redundancy
This adds the following line to the SDP:
a=rtpmap:someid RED/48000/2
To test start Chrome with
--force-fieldtrials=WebRTC-Audio-Red-For-Opus/Enabled
BUG=webrtc:11640
Change-Id: I8fa9fb07d03db5f90cdb08765baaa03d3d0458cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176372
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31562}
This interface has a couple of issues. Primarily for me, it makes it
difficult work with the paced sender as we need to either temporarily
release a lock or force a thread-handover in order to avoid a cyclic
lock order.
For video in particular, its behavior is also falky since header sizes
can vary not only form frame to frame, but from packet to packet within
a frame (e.g. TimingInfo extension is only on the last packet, if set).
On bitrate allocation, the last reported value is picked, leading to
timing issues affecting the bitrate set.
This CL removes the callback interface and instead we simply poll the
RTP module for a packet overhead. This consists of an expected overhead
based on which non-volatile header extensions are registered (so for
instance AbsoluteCaptureTime is disregarded since it's only populated
once per second). The overhead estimation is a little less accurate but
instead simpler and deterministic.
Bug: webrtc:10809
Change-Id: I2c3d3fcca6ad35704c4c1b6b9e0a39227aada1ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173704
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31185}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
This makes an assumption that if we have variable frame length then we
will increase payload bitrate up to priority bitrate before adapting the
frame length.
Bug: webrtc:11001
Change-Id: Iec51d5ccce053d55ccd30a9e4712765227e10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169852
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30713}
This fixes two cases when the allocation is not updated correctly:
- The frame length range is not updated when audio network adaptor is enabled.
- The per-packet overhead is not updated unless the bitrate observer has been reconfigured.
Bug: webrtc:11001
Change-Id: I2ee25f956741a4be08661f874556582dd60a3bd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169848
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30709}
This ensures that overhead calculation is correct by default when
enabling the WebRTC-SendSideBwe-WithOverhead field trial.
We keep the legacy mode to allow downstream projects already relying on
WebRTC-SendSideBwe-WithOverhead to preserve the current behavior.
Bug: webrtc:6762
Change-Id: I84369c760d59345a48ec352997dbed6d2db21d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167862
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30424}
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}
