Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by
- moving getters for EncodedImage fields up to EncodedImage
- copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame
- Removing EncodedFrame's inheritance of VCMEncodedFrame
We leave VCMEncodedFrame as part of the (near) deprecated
VideoCodingModule code. The only place which needs to accept either is
in the generic decoder.
Bug: webrtc:9378, b:296992877
Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40639}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.
The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.
Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262
WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.
This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.
Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
Remove code where integer -1 as delay is used to represent unset value.
Bug: webrtc:13756
Change-Id: I16a01e12c25a09ce21a971c9edabf47af5936662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316923
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40592}
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.
Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta
Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.
Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
which is present if a fec mechanism like FlexFEC is negotiated
spec change:
https://github.com/w3c/webrtc-stats/pull/765
BUG=webrtc:15250
Change-Id: I7d71d49fab0153d734f22831e6684d2acfc647fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314981
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40514}
This patch adds a StunDictionary.
The dictionary has a reader and a writer.
A writer can update a reader by creating a delta.
The delta is applied by the reader, and the ack is applied by the
writer.
Using this mechanism, two ice agents can (in the future) communicate
properties w/o manually needing to add new code.
The delta and delta-ack attributes has been allocated at IANA.
Bug: webrtc:15392
Change-Id: Icdbaf157004258b26ffa0c1f922e083b1ed23899
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40513}
This is part of the long term plan to stop using pointer + length
to pass around buffers.
Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
since this may contain sensitive data, just like the address.
BUG=None
Change-Id: I3faa1512a15467cd5cc4bcbcaebadb736f1bae07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40473}
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.
Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
The existing equality check did not always work since content_type
is sometimes overloaded with extra internal information such as simulcast layer index. Fix by using the videocontenttypehelpers::IsScreenshare helper method.
Bug: webrtc:15381
Change-Id: I2fe84e7f036ea2c223e4fa6dd58af1c4c0bcfbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312261
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40448}
This CL cleans up all local conversions, in favor of the common helper
function.
Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
This avoids the risk of a higher level owner approving something getting
added to the getStats() API without +1s from stats owners.
Bug: None
Change-Id: Iedd7133d0e943d1db6977dc0e1d406e5b545e31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311543
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40412}
Move the SetRTPTimestamp method from TransformableAudioFrameInterface
to the base class, so that RTPTimestamps can also be modified on encoded
video frames.
Bug: webrtc:14709
Change-Id: I355be527c2be201c9201e04c431394c962237140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310781
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40378}
Instead switch to specific getters, or methods only defined on specific implementations rather than part of the public API.
Once uses are removed from Chromium, I'll mark GetHeader() deprecated
and eventually remove it.
Bug: chromium:1456628
Change-Id: I19b80489b3a0322c201e24994494cfbb742ee13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309780
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40344}
specified in https://github.com/w3c/webrtc-stats/pull/762
and take FlexFEC into account for receive statistics.
BUG=webrtc:15250
Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40325}
Scoped enums do not get automatically promoted to their underlying type,
so these uses have undefined behavior and Clang recently started warning
about it.
Bug: chromium:1456289
Change-Id: I9cf4e5a68378930a3bf7d8ac7b0a21eaf0d12670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309520
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40321}
Introduces a field trial
WebRTC-IncreaseIceCandidatePriorityHostSrflx
that adjusts the priority of non-relay candidates such that the STUN priority attribute calculated as
(prflx-type-preference << 24) | (priority & 0x00FFFFFF)
as described in
https://www.rfc-editor.org/rfc/rfc5245#section-7.1.2.1
will satisfy the condition that the STUN priority of server-reflexive candidates will always be higher than the STUN priority of relay candidates.
Previously this was not the case because the TURN relay preference was added to the local_preference for relay candidates, making it higher than the local_preference of srflx candidates gathered from the same interface.
This led to cases where the resulting candidate pair priority of a srflx-relay pair was higher than the candidate pair priority of a srflx-srflx pair, i.e. using a TURN server in cases that work using a direct P2P connection.
Whether the field trial is active can be observed by checking that
priority-of-srflx-candidate&0x00FFFFFF
is greater than
priority-of-relay-candidate&0x00FFFFFF
BUG=webrtc:13195,webrtc:5813,webrtc:15020
Change-Id: Ib91708fbe7310b6454f93158a45c9d77da009091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292700
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40311}
Make outgoing encoded audio frames inherit from the same Audio interface
that incoming frames inherit from, to align them and make it possible to
eg clone frames regardless of their direction.
Also begin removing GetHeader() from the Audio interface, replacing it
with getters for the specific values we actually need to propagate in
the API: sequence number and CSRCs. This makes it much easier to treat
incoming and outgoing frames the same, even if they don't have full
RtpHeaders prepared at the point of the transform.
Bug: chromium:1453226
Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40309}
This is no longer needed after downstream redefinitions are deleted.
Bug: webrtc:15241
Change-Id: Iea6839bff781fe7d0c56b4739f3d43398c70f2b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40306}
This is a reland of commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199
Downstream project has been updated.
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: Ibc7aeb518ed0bd7f1d725f140132c99e5a89bcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308880
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40305}
This reverts commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199.
Reason for revert: Breaks downstream project
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: I05db80ba9f29460239de82cea9d95136e4c708e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308860
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40298}
Also add a preprocessor definition to avoid redefinition in downstream projects.
Bug: webrtc:15241
Change-Id: Ic55d98c3d3a69b9b19195ee78f03af6e38fdd0e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40289}
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
with a single encoding or multiple encodings. As long as only one
encoding is active we get a single L1Tx ssrc, same as legacy API.
Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.
The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.
This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.
Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
This can be done now as the function SetRTPTimestamp is now overriden
in blink MockTransformableAudioFrame.
Change-Id: I4fa4cb81d0282fea864818f0f2d9a5ed881a5d30
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40257}
This CL implements {,Logging}DelayVariationCalculator, whose purpose is to calculate simple inter-arrival metrics for a sequence of RTP frames. Uses could include RtcEventLog analysis and ad hoc testing.
Want lgtm: asapersson
Bug: webrtc:15213
Change-Id: I3f9d13a2c4fa66b6f1229c1b6fcd66a6911070de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306741
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40247}
This change will make it possible to let us modify timestamp in
RTCEncodedAudioFrame.
Change-Id: I97e9571c258fd718d6c211014f1476ca46c78097
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307501
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40238}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Currently FecController knows about network conditions, these information can be used to control RTX settings in-call.
Change-Id: I8f84164aeac48ea13b7f1cf82fd7424431f98ada
Bug: webrtc:15167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304800
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40192}
Add an API to pass AudioFrameProcessor as a unique_ptr.
Bug: webrtc:15111
Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40187}
