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}
Elapsed time since last played out frame could be incorrect in GetCurrentEstimatedPlayoutNtpTimestampMs (e.g. if playout stops).
Bug: webrtc:7065
Change-Id: Ibb40b153ea7291e2cd3843c71ab44ff0fb00971c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168720
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30604}
This reverts commit 0e96535be97916d8fcaa9873ffab3c636539f9d8.
Reason for revert: Downstream test failure
Original change's description:
> Inlines NullAudioPoller functionality into AudioState class.
>
> As part of this, we also use TaskQueue and RepeatedTask rather
> than rtc::Thread + rtc::MessageHandler. With the ultimate goal of
> deprecating rtc::Thread.
>
> Bug: webrtc:9883
> Change-Id: I2fb851ac31ee2431435d51de78ff446572512201
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167528
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30430}
TBR=saza@webrtc.org,srte@webrtc.org
Change-Id: I4c77259f7b6477fc1cb79350f2d47817f106770d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168046
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30431}
As part of this, we also use TaskQueue and RepeatedTask rather
than rtc::Thread + rtc::MessageHandler. With the ultimate goal of
deprecating rtc::Thread.
Bug: webrtc:9883
Change-Id: I2fb851ac31ee2431435d51de78ff446572512201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167528
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30430}
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}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This reduces indirection and makes it easier to follow code. It also
fits into a long term strategy of reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: I2661c4aa6c561f7691beaaa289636254f7a58b72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166042
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30273}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
This CL introduces a channel remapping for the mono input case that
is more tailored to the VoIP usecase.
The CL contains a kill-switch that can be used to fall back to the old
mapping behavior in case a need for this is perceived.
Bug: chromium:1027117
Change-Id: Idaaba6eac952e6436beaaf5a1a697cfab8f63286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160280
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29877}
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.
Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
Injecting both a custom NetEqFactory and an AudioDecoderFactory is not
supported, in that case the AudioDecoderFactory should be wrapped inside
the NetEqFactory.
Bug: webrtc:11005
Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29673}
This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.
This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.
Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
Fixes an issue with NullAudioPoller calling the mixer every 10 ms when
no call is ongoing.
Bug: b/142775365
Change-Id: I77eeddadaf08b358cce2b389c70e4f2baf1d5627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157176
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29508}
This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.
Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.
This prepares for reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
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@{#29490}
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.
Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
Well, in fact we need to return both. But return codec sample rate
separately and let the SdpAudioFormat contain the RTP clockrate,
otherwise we're essentially lying to our callers.
Bug: webrtc:11028
Change-Id: I40f36cb9db6b9824404ade6b0515a8312ff97009
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156307
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29444}
