This allows users to inject the residual echo detector, as a step toward making it an optional part of compilation.
Bug: webrtc:11292, webrtc:11539
Change-Id: I7fcc8dbaced67a82851cd6cdcbc115eb01c21fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174040
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31222}
Add peer's name to params and use it for logging and metrics naming
for whole peer related metrics.
Bug: webrtc:11479
Change-Id: Ia7e3fc4839c90a958d66910614515ac02a96e389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174752
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31215}
Modernise functions to unified MOCK_METHOD macro,
delete few deprecated functions on the way.
add one missing function (in MockEncodedImageCallback)
Remove proxy mock function (in MockVideoBitrateAllocatorFactory)
Remove default constructors and destructors
Bug: None
Change-Id: Ibebb0d9e3c9be5877649af7bde8b87222ddf04fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174751
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31195}
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.
Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
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}
After the migration to passing frame video source implementation directly, part of the peer connection framework code became redundant. Removing screen_share_config and capturing_device_index from the VideoConfig is to be done in later reviews.
Bug: webrtc:11534
Change-Id: I7a8ea85d26d00fb5bfe7ec0d2facef9c44a0f749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174541
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31178}
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
This is one last CL that includes the rest of VoIP API implementation.
Bug: webrtc:11251
Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31168}
In order for WebRTC to be able to include packet overhead in its
bitrate calculations, the AudioEncoder::GetFrameLengthRange()
function must be implemented by all audio encoders. Making this
member function pure virtual as per the following PSA:
https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0
Bug: webrtc:11427
Change-Id: I30d297ef05f57453bfc257624729559057cad118
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31127}
Replaces this with 2 methods instead, adding clarity.
ClearAdaptationStats
- Resets the adaptations statistics to 0. This is done,
when the degredation is reset, for example when the preference
is changed to/from BALANCED.
UpdateAdaptationMaskingSettings
- Updates the settings for adaptation statistics reporting.
This way we don't report quality adaptations if quality scaling
is not enabled (same for resolution/fps scaling).
The adaptation counting inside the SendStatisticsProxy is
now done in a struct that counts the totals, and then masks
out these counts based on the adaptation settings. The
MaskedAdaptationSteps uses optionals to hide the values we
shoudn't report, while the AdaptationSteps always hold the real
totals.
All tests have been updated to use the Reset/Clear method as needed.
Now that AdaptationCounters and AdaptSteps use the same structure,
AdaptationCounters was moved to api/video and replaces AdaptSteps.
The AdaptReason enum is also redundant now, and will be removed
in a follow-up CL.
R=hbos@webrtc.org
Bug: webrtc:11392
Change-Id: Iaed6488581325d341a056b5bbf76a01c19d6c282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171685
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31083}
Add new methods in the FrameTransformerInterfaces, passing the ssrc on
registering the transformed frame callback, to associate separate frame
transformer sinks for each ssrc. Same for unregister.
Bug: chromium:1065838
Change-Id: I8a406815e9d0cce5199f9df06c286d8b10d75b4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173183
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31047}
Add a new frame interface to be used by frame transformers in Insertable
Streams. TransformableFrameInterface will replace
video_coding::EncodedFrame in a follow up CL, once downstream
dependecies are updated to use the new interface.
Until the functions using video_coding::EncodedFrame are removed from
the API, the video sender and receiver frame transformer delegates call
both function versions to avoid breaking tests downstream.
The TransformableFrameInterface will be used for both audio and video
frame transformers in follow-up CLs.
Bug: webrtc:11380
Change-Id: I9389a8549c156e13b1d8c938ff51eaa69c502f33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171863
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30941}
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
If a STUN attribute is in the "comprehension-required" range
(0x0000-0x7FFF), and the implementation does not recognize it, this
should be treated as an error (as per RFC5389), with different behavior
depending on the type of the message received.
Bug: webrtc:9063
Change-Id: Ic31b0cdd3c26772c21d770b44fe4ee4a1b47030a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/64500
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30925}
To allow construct that object from an existent string_view without explicit conversion
Bug: webrtc:11428
Change-Id: I38d93573be72e307bdf7068a6300d10cf46d2d62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171689
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30904}
To migrate on new GetStats API and properly support target encode bitrate
for regular, simulcast and svc cases we need to calculate it inside video
quality analyzer getting values from SetRates in VideoEncoder.
Bug: webrtc:11381
Change-Id: Ia37acac764ed3c30f64cdbfda8906d543fa03ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171501
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30881}
This CL renames the shadow filter in AEC3 to have the more accurate name
coarse filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "shadow" with "coarse" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: I28d6041d0d34e85f8f8048d004b44a1a5f07bb07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170981
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30846}
This CL replaces a few usages of const std::string& with
absl::string_view, to comply closer with
https://abseil.io/tips/1.
Bug: webrtc:11428
Change-Id: Ibf6fac9b084cb21e17db63f73d667793ab9cafeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170466
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30845}
This CL renames the main filter in AEC3 to have the more accurate name
refined filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "main" with "refined" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: Ifd0aab34e291736a2250e0986348404618630b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170825
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30843}
