These are no longer needed since the RTP transport runs on the worker
thread now.
Some tests that were too strict on ordering needed change.
Bug: none
Change-Id: I4265cb1a4fd3355208f19aefdbb7abeb45b6cadf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335700
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41599}
Split from
https://webrtc-review.googlesource.com/c/src/+/318283
to reduce CL size. Takes a different and (hopefully) simpler
approach.
BUG=webrtc:15579
Change-Id: I8517ffbeb0f0a76db80e3e367de727fb6976211d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325023
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41073}
NetworkLinkRtcpObserver is similar to RtcpBandwidthObserver but pass
time variables using unit types instead of raw integers.
Bug: webrtc:13757
Change-Id: Iaa0bbe0b108620b3a24013c40e7d9004032e904d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305022
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40087}
Delete VoERtcpObserver proxy:
pass BWE related message directly to transport controller
pass ReportBlock directly to ChannelSend, assuming there will be single report block per source ssrc
Bug: None
Change-Id: I8378326bff1dc3c2736960166fc782ee822a9c12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305224
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40081}
ReportBlockData class is better documented and has wider usage.
Bug: webrtc:13757
Change-Id: Ie5f2275f2f0236267172e6dd1ce5c2dfb2193ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304101
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39980}
This reverts commit dd557fdb1e.
Reason for revert: Looks like the Chromium FYI builders are failing.
Original change's description:
> [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream
>
> This remove use of MaybeWorkerThread* rtp_transport_queue_ from
> AudioSendStream. The worker queue is alwauys assumed ot be used where
> rtp_transport_queue_ was used.
>
> Bug: webrtc:14502
> Change-Id: Ia516ce7340d712671e0ecb301bba9d66e7216973
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300400
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39816}
Bug: webrtc:14502
Change-Id: I0547548032756fc579b76b6bb362f576aa06b8f7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39820}
This remove use of MaybeWorkerThread* rtp_transport_queue_ from
AudioSendStream. The worker queue is alwauys assumed ot be used where
rtp_transport_queue_ was used.
Bug: webrtc:14502
Change-Id: Ia516ce7340d712671e0ecb301bba9d66e7216973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300400
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39816}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
This spills to a few more clasess....
Change-Id: Iea79e3b4ac86b30db6f13da89a47ab7000c5440a
Bug: webrtc:14502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277803
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38334}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
which do not make sense for audio due to lack of support for RTX.
BUG=webrtc:13279
Change-Id: Ida42d8912bf993f01e0dc5c6ffbdbf4b84495c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235061
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35309}
This reverts commit 2c41cbae37.
Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2.
Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c05.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34897}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34930}
This reverts commit fb0dca6c05.
Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=hta,hbos,minyue
Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
This will soon become a compile-time error. Fix class hierarchies that
wrap StrictMock in a NiceMock or vice-versa by removing redundant
wrappings and removing inheritance from Nice/StrictMock and fixing the
call sites as appropriate.
Bug: b/173702213
Change-Id: Ic90b1f270c180f7308f40e52e358a8f6a6baad86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32783}
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 a135557b3c
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 a135557b3c.
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}
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}
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}
As of this CL:
https://webrtc-review.googlesource.com/c/src/+/173704
...we now call AudioEncoder::OnReceivedOverhead() too often, since we
don't check if overhead has actually changed.
This CL rectifies that.
Bug: webrtc:10809
Change-Id: I0b86e0296a7860dde3e62e817f9b941fa82afe4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175009
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31279}
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}
The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in ChannelSend, where the frame
transformation will occur in the follow-up CL.
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30946}
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 is a reland of 086055d0fd
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 8c79c6e1af
>
> 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 8c79c6e1af
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 8c79c6e1af.
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}
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}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
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}
