Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/c/105180
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25130}
In this CL the use of the stationarity properties at init is set to true by default.
Bug: webrtc:9865, chromium:894439
Change-Id: I716ce0d792a50616dc38cc0ba6f2c702549a81cc
Reviewed-on: https://webrtc-review.googlesource.com/c/105303
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25123}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This CL replaces CHECKs and crashes with DCHECKs and default values.
Bug: webrtc:9535
Change-Id: Ib4b16421699c633d0e9ef140189861c8179450f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105003
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25111}
Basic integration of media_transport in JSepTransportController.
- Creates media_transport if media transport factory provided in jsep transport controller configuration.
- Unittest that makes sure media_transport is created with correct caller or callee setting.
- Added fake_media_transport, for now simple implementation which only stores caller/callee, but in the future fake media transport will be expanded to pass frames between two fake media_transports, which will enable audio / video integration tests.
NEXT STEPS: Once integration of media_transport with PeerConnection (https://webrtc-review.googlesource.com/c/src/+/103860) lands, we can start passing media transport factory from peer connection to jsep transport controller.
NOTE: Includes missing include change from https://webrtc-review.googlesource.com/c/src/+/103540 (otherwise this change will not compile)
Bug: webrtc:9719
Change-Id: I1e8a521beab445aa9f7ea93c8d7a537dc137d11c
Reviewed-on: https://webrtc-review.googlesource.com/c/104400
Commit-Queue: Anton Sukhanov <sukhanov@google.com>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25096}
If the need arises, please use:
using std::swap;
swap(a, b);
which falls back to a generic std::swap.
Bug: webrtc:9855
Change-Id: I819839d160fc7ae289310a13e3988cdb3f0b3086
Reviewed-on: https://webrtc-review.googlesource.com/c/104100
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25084}
In order to correctly implement RTCPeerConnectionState and RTCIceConnectionState the ice transports need to support RTCIceTransportState.
This CL adds an implementation parallel to the current non-standard IceTransportState. It's not currently used anywhere. The old implementation will remain in place until we're ready to switch RTCIceConnectionState over.
Bug: webrtc:9308
Change-Id: I30e2bbb5b4fafa410261bcd9d5e3b76c03435feb
Reviewed-on: https://webrtc-review.googlesource.com/c/103220
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25078}
This CL changes the tuning of the echo suppressor for the case when
there is echo only. The resulting effect is a slight increase of
transparency
Bug: webrtc:9844,chromium:893744
Change-Id: I5e6a867e0d03dc3a468a8f5cfa64103e001baae1
Reviewed-on: https://webrtc-review.googlesource.com/c/104760
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25075}
This CL utilizes the AEC3 ability to tailor the suppressor during
situations when the nearend dominates over the residual echo. This is
done by increasing the thresholds for transparent echo suppressor
behavior when the nearend is strong compared to the residual echo.
Bug: webrtc:9836, chromium:893744
Change-Id: Ic06569eefc7f2557b401db43b3ac84b299071294
Reviewed-on: https://webrtc-review.googlesource.com/c/104460
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25071}
This CL adds flags to the PacketOptions and PacktInfo struct that are
intended to be used to indicate if the packet belongs to a media stream
that is part of bitrate allocation as well as if it is included in
transport wide packet feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303
Reviewed-on: https://webrtc-review.googlesource.com/c/104920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25069}
This CL adds fields to packet feedback structs to indicate the amount of
data that was sent prior to the represented packet without being part
packet feedback, but part of bitrate allocation.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: I716a5325e2b7022ba6b3f90653542caafb056793
Reviewed-on: https://webrtc-review.googlesource.com/c/104921
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25067}
This is a new attempt to reduce the filter divergence
during double-talk without regressing in clock-drift
scenarios.
- The error_floor in decreased to allow for slow adaptation
when the filter performs well.
- The leakage_diverged is increased to allow for fast adaptation
when the shadow filter performs better.
- A new parameter, error_ceil, was added to stop the filter from
adapting too fast.
Bug: webrtc:9746,chromium:883264
Change-Id: Ie2868d2388b48412a192a004ec13f9eff34517b8
Reviewed-on: https://webrtc-review.googlesource.com/c/100460
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25063}
This CL lowers the default reverb decay to better match the standard
rooms where calls are made.
Bug: webrtc:9843
Change-Id: I46f1a629ecfdd72561829326d4fa58ede8107b6c
Reviewed-on: https://webrtc-review.googlesource.com/c/104740
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25061}
And use RTCConfiguration to enable/disable it on a per connection basis.
With the advent of MediaTransportInterface, we need to be able to enable
it on the per PeerConnection basis.
At this point PeerConnection will not take any action when the
MediaTransportInterface is set; this code will land a bit later, and
will be accompanied by the tests that verify correct setup (hence no tests right now).
At this point this is just a method stub to enable further development.
Bug: webrtc:9719
Change-Id: I1f77d650cb03bf1191aa0b35669cd32f1b68446f
Reviewed-on: https://webrtc-review.googlesource.com/c/103860
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25053}
It is done to better show what for this class exists and also restore
correspondence between config and interface, that is implemented by
configurable object.
Bug: webrtc:9630
Change-Id: I28456d1c792d67d9b2a405c8599054137a5d596a
Reviewed-on: https://webrtc-review.googlesource.com/c/104003
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25041}
Add FakeVp8Decoder that parse width and height from the payload.
Add unit test for testing that width and height is set when decoding frames.
Bug: none
Change-Id: Ifbfff4f62f99625309ce0ef21cf89c76448769d8
Reviewed-on: https://webrtc-review.googlesource.com/c/103140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25038}
This is quite useful in many places where we need to restrict the range
of a DataRate. It makes it easier to read the intention than with:
value_ = std::max(some_lower_limit, std::min(value_, some_upper_limit));
The naming follows the naming for rtc::SafeClamp.
Bug: webrtc:9709
Change-Id: I08e05197acec325d85babd2a06806a8667f2fcb1
Reviewed-on: https://webrtc-review.googlesource.com/c/104040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25023}
Automatic detection if one-byte header or two-byte header should be used based
on extension ID and extension length.
Bug: webrtc:7990
Change-Id: I9fc848ecc59458d1ca97bace0e57ea04d3d0ced6
Reviewed-on: https://webrtc-review.googlesource.com/c/103422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25004}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.
If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.
Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}
This name will better describe what implementation should do and how
users will interact with this class. The real simulation is done
by FakeNetworkPipe and this class is just operates with packages
metadata, so it is more about describing behavior.
Bug: webrtc:9630
Change-Id: I00977e6be0ca84e7c233b4c35f0677e8263e4382
Reviewed-on: https://webrtc-review.googlesource.com/c/95944
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24989}
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.
Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24985}
This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53.
Reason for revert: Breaks chromium.webrtc.fyi bots.
Original change's description:
> Export symbols needed by the Chromium component build (part 1).
>
> This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
> to mark WebRTC symbols as visible from a shared library, this doesn't
> mean these symbols are part of the public API (please continue to refer
> to [1] for info about what is considered public WebRTC API).
>
> [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
>
> Bug: webrtc:9419
> Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24969}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/103720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24974}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
Reviewed-on: https://webrtc-review.googlesource.com/c/103505
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24969}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.
Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.
Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24917}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
