I've updated all the tests that previously were calling this method on
the wrong thread, so we can enable this check now.
I've also landed some changes that simplify the threading model in this
class and subsequently I've removed some locks and can remove some more
in this CL.
Added some comments about future improvements for GetStats() to reduce
synchronization.
Simplified CallStats::OnRttUpdate() to have one fewer async methods.
Bug: webrtc:10847
Change-Id: I48e6809172142cc4be4385b7d4aa2affb52a963a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148588
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28821}
There aren't any tests for this and the code isn't currently
active except for the fact that it adds complexity to the Call
class, synchronization into the active code path and makes future
improvements to the class more complex or impossible.
Bug: webrtc:9719
Change-Id: Ia41af0b2186b8a36ca70a07858990b6af7f3a5c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148078
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28807}
Also removing the stop_event_ from the RampUpTester class, which I missed in review 148067.
Bug: webrtc:10847
Change-Id: I102cc75287503915b51e37ea4ee01dfcc2437699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148062
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28801}
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
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}
In some situation, we disable simulcast in the encoder pipeline without
changing the sender's RIDs and it would crash.
This should only happen now when requesting simulcast with VP9 codec,
for which you currently get SVC instead.
Bug: webrtc:10660
Change-Id: I4f3b3d7760aded8f0769f8357c03ed8580ea46fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145336
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28669}
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.
The functionality is negotiated using SDP.
It's added with a field trial that allow disabling the functionality in
case there's any issues.
Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
Implements RTCAudioSourceStats members:
- audioLevel
- totalAudioEnergy
- totalSamplesDuration
In this CL description these are collectively referred to as the audio
levels.
The audio levels are removed from sending "track" stats (in Chrome,
these are now reported as undefined instead of 0).
Background:
For sending tracks, audio levels were always reported as 0 in Chrome
(https://crbug.com/736403), while audio levels were correctly reported
for receiving tracks. This problem affected the standard getStats() but
not the legacy getStats(), blocking some people from migrating. This
was likely not a problem in native third_party/webrtc code because the
delivery of audio frames from device to send-stream uses a different
code path outside of chromium.
A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the
send-side audio levels to the RTCAudioSourceStats, while keeping the
receive-side audio levels on the "track" stats. This allows an
implementation to report the audio levels even if samples are not sent
onto the network (such as if an ICE connection has not been established
yet), reflecting some of the current implementation.
Changes:
1. Audio levels are added to RTCAudioSourceStats. Send-side audio
"track" stats are left undefined. Receive-side audio "track" stats
are not changed in this CL and continue to work.
2. Audio level computation is moved from the AudioState and
AudioTransportImpl to the AudioSendStream. This is because a) the
AudioTransportImpl::RecordedDataIsAvailable() code path is not
exercised in chromium, and b) audio levels should, per-spec, not be
calculated on a per-call basis, for which the AudioState is defined.
3. The audio level computation is now performed in
AudioSendStream::SendAudioData(), a code path used by both native
and chromium code.
4. Comments are added to document behavior of existing code, such as
AudioLevel and AudioSendStream::SendAudioData().
Note:
In this CL, just like before this CL, audio level is only calculated
after an AudioSendStream has been created. This means that before an
O/A negotiation, audio levels are unavailable.
According to spec, if we have an audio source, we should have audio
levels. An immediate solution to this would have been to calculate the
audio level at pc/rtp_sender.cc. The problem is that the
LocalAudioSinkAdapter::OnData() code path, while exercised in chromium,
is not exercised in native code. The issue of calculating audio levels
on a per-source bases rather than on a per-send stream basis is left to
https://crbug.com/webrtc/10771, an existing "media-source" bug.
This CL can be verified manually in Chrome at:
https://codepen.io/anon/pen/vqRGyq
Bug: chromium:736403, webrtc:10771
Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28480}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28385}
This moves the conversion from RtpPacketReceived to ReceivedPacket to
Call rather than RtpTransportController. This prepares for reusing the
struct for receive side network state estimation.
Bug: webrtc:10742
Change-Id: I9581438bc912ef4bb635a5d9a6dea488cf871d48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141872
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28284}
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
Zero bitrate caused division by zero in DCHECK for max bitrate.
Added unit tests to ensure that setting zero bitrate does not crash.
> Original change's description:
> > Cleanup of video packet overhead calculation.
> >
> > This CL updates the video packet overhead calculation to make it more
> > clear. This prepares for future work on improving the accuracy of the
> > calculation.
> >
> > Bug: webrtc:9883
> > Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28040}
Bug: webrtc:10674
Change-Id: I156d1ee5546ede7e43ae1d9a298dcaba6071230f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140890
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28212}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
This reverts commit 35d4e43f169e7cb237bce9501db29ea4b69820cd.
Reason for revert: Breaks downstream.
Original change's description:
> Reland "Cleanup of video packet overhead calculation."
>
> This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
>
> The calculation was rewritten using the new Frequency type to
> avoid the division by zero error introduced by the previous CL.
>
> Original change's description:
> > Cleanup of video packet overhead calculation.
> >
> > This CL updates the video packet overhead calculation to make it more
> > clear. This prepares for future work on improving the accuracy of the
> > calculation.
> >
> > Bug: webrtc:9883
> > Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28040}
>
> Bug: webrtc:10674
> Change-Id: Ib5cb6f05cfa7d097f89ac6fdcf198f2fc1b26b58
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138219
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28194}
TBR=nisse@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: Ib6c3c123590b815c4be12966cdae02f91c61ab34
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10674
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140889
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28195}
This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
The calculation was rewritten using the new Frequency type to
avoid the division by zero error introduced by the previous CL.
Original change's description:
> Cleanup of video packet overhead calculation.
>
> This CL updates the video packet overhead calculation to make it more
> clear. This prepares for future work on improving the accuracy of the
> calculation.
>
> Bug: webrtc:9883
> Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28040}
Bug: webrtc:10674
Change-Id: Ib5cb6f05cfa7d097f89ac6fdcf198f2fc1b26b58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138219
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28194}
Using this flag, an encoder may inform the RTP sender module that
the packet is not elligible for retransmission. Specifically, it
may not be retransmitted in response to a NACK message,
nor because of early loss detection (see CL #135881).
Bug: webrtc:10702
Change-Id: Ib6a9cc361cf10ea7214cf672e05940c27899a6be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140105
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28169}
