For the automatic content type detection we need to know if the update
rect is trusted or just not available.
Currently we only care if it's not empty, so in case of no update rect
available, full frame resolution was set as a changed region.
This CL makes the update_rect field optional but should be a no-op in the
current code, as absence of update_rect is treated as a full update via
a new getter method |update_rect_or_full_frame()|.
Bug: webrtc:11058
Change-Id: I913545b71ac2fc845861549ac34eb1b630012109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158673
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29654}
The deleted members are bytes_sent, bytes_received,
and (receive-side) fraction_lost.
Followup to https://webrtc-review.googlesource.com/c/src/+/143178.
Bug: webrtc:10744, webrtc:10525
Change-Id: Ibd8b15619c78b82a9dc18abddb6e1687fae60c88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143902
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29566}
This enables creation and removal of receive streams with SSRC 0.
Several related methods, for example SetOutputVolume, still use 0 as a
special value.
Bug: webrtc:8694
Change-Id: I341e6bd6c981c9838997510d8d712ad2948f6460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29398}
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.
The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.
Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.
Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
It's propagated from ReceiveStatistics up to VoiceReceiverInfo,
and then not used. It's not part of the standard stats.
Bug: None
Change-Id: I90ce6a72e3ca846adbbba5d3023fef18a2169018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149164
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28933}
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels. There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.
PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.
Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks. This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state. This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.
For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.
Datagram transport use is negotiated. As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer. If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport. If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.
If DTLS is not enabled, there is no viable fallback. In this case, the data
channels are negotiated as media transport data channels. If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.
Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
This is a reland of 2b9fa09fa3e3379fd8e76490c394f25670352ef2.
It got reverted because I forgot to whitelist the new metrics in chromium,
which has now been done:
https://chromium-review.googlesource.com/c/chromium/src/+/1760209
Relanding requires no changes to the CL.
Original change's description:
> [GetStats] Expose video codec implementation in standardized metrics.
>
> Spec issue: https://github.com/w3c/webrtc-stats/issues/445
> Spec PR: https://github.com/w3c/webrtc-stats/pull/473
>
> Now that the spec's RTCCodecStats.implementation has moved to
> RTCOutboundRtpStreamStats.encoderImplementation and
> RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
> using the same string that the legacy getStats() API used.
>
> Bug: webrtc:10890
> Change-Id: Ic43ce44735453626791959df3061ee253356015a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28877}
TBR=ilnik@webrtc.org
Bug: webrtc:10890
Change-Id: Ib874b608856c2795b1ca08f6af43c61dd859ea21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149800
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28887}
This reverts commit 2b9fa09fa3e3379fd8e76490c394f25670352ef2.
Reason for revert: speculative revert since it seems to break Chrome FYI bots. See https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4206
Original change's description:
> [GetStats] Expose video codec implementation in standardized metrics.
>
> Spec issue: https://github.com/w3c/webrtc-stats/issues/445
> Spec PR: https://github.com/w3c/webrtc-stats/pull/473
>
> Now that the spec's RTCCodecStats.implementation has moved to
> RTCOutboundRtpStreamStats.encoderImplementation and
> RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
> using the same string that the legacy getStats() API used.
>
> Bug: webrtc:10890
> Change-Id: Ic43ce44735453626791959df3061ee253356015a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28877}
TBR=ilnik@webrtc.org,hbos@webrtc.org
Change-Id: Ia0b7f9806564cf28881c50d6371b8141a22e3431
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10890
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149175
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28879}
Spec issue: https://github.com/w3c/webrtc-stats/issues/445
Spec PR: https://github.com/w3c/webrtc-stats/pull/473
Now that the spec's RTCCodecStats.implementation has moved to
RTCOutboundRtpStreamStats.encoderImplementation and
RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
using the same string that the legacy getStats() API used.
Bug: webrtc:10890
Change-Id: Ic43ce44735453626791959df3061ee253356015a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28877}
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}
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}
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}
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 change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
In several places VideoFrame::Builder is used to create a new VideoFrame
when intent is to change only one or two fields of a const VideoFrame&.
This approach is bad because each and every metadata field have to be
added to all the places.
Instead, this CL adds missing setters and refactors the code to use
full copy of a VideoFrame and update required fields only.
Along the way few actual bugs are fixed, e.g. when ColorSpace isn't copied
when frame rotation or buffer is cropped or converted.
Bug: webrtc:10460
Change-Id: I2895a473ca938b150eed2916c689060bdf58cb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140102
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28170}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
H264 requires that the packetization modes are the same in order to
consider the code the same. This logic was added to VideoCodec::Matches
but was not reflected in IsSameCodec. This could manifest itself when a
remote description with an unsupported packetization mode is set.
Bug: webrtc:10693
Change-Id: Icda07f7d56a464895d2267a41cc0f2fd9d5f42ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138983
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28126}
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats
In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
video streams, but the code is not wired up to make use of these stats.
In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.
[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067
Bug: webrtc:10456
Change-Id: Icd20452cb4b4908203b28ae9d9f52c25693cf91d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138065
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28071}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.
TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.
Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}
This is a standardized metric:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
We estimate the target frame size in bytes from the current encoder
target bitrate and encoder framerate.
We would expect that the average bytes produced by the encoder would
over time match the average target, which is calculated by polling
getStats() twice and dividing the delta totalEncodedBytesTarget with
the delta framesEncoded. This is meant to make googTargetEncBitrate
obsolete.
Bug: webrtc:10446
Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28022}
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().
Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27979}
This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1.
Reason for revert: Tightened protocol name handling.
Original change's description:
> Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
>
> This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.
>
> Reason for revert: fuzzer failures
>
> Original change's description:
> > Reland "Version 2 "Refactoring DataContentDescription class""
> >
> > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
> >
> > Original change's description:
> > > Version 2 "Refactoring DataContentDescription class"
> > >
> > > (substantial changes since version 1)
> > >
> > > This CL splits the cricket::DataContentDescription class into
> > > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > > and cricket::SctpDataContentDescription (used for SCTP only).
> > >
> > > SctpDataContentDescription no longer inherits from
> > > MediaContentDescriptionImpl, and no longer contains "codecs".
> > >
> > > Due to usage of internal interfaces by consumers, shimming the old
> > > DataContentDescription API is needed.
> > >
> > > A new cricket::DataContentDescription class is defined, which is
> > > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > > It exposes as little functionality as possible, but supports the
> > > concerned consumer's usage
> > >
> > > Design document:
> > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> > >
> > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > >
Bug: webrtc:10358
Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27933}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.
Reason for revert: fuzzer failures
Original change's description:
> Reland "Version 2 "Refactoring DataContentDescription class""
>
> This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
>
> Original change's description:
> > Version 2 "Refactoring DataContentDescription class"
> >
> > (substantial changes since version 1)
> >
> > This CL splits the cricket::DataContentDescription class into
> > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > and cricket::SctpDataContentDescription (used for SCTP only).
> >
> > SctpDataContentDescription no longer inherits from
> > MediaContentDescriptionImpl, and no longer contains "codecs".
> >
> > Due to usage of internal interfaces by consumers, shimming the old
> > DataContentDescription API is needed.
> >
> > A new cricket::DataContentDescription class is defined, which is
> > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > It exposes as little functionality as possible, but supports the
> > concerned consumer's usage
> >
> > Design document:
> > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> >
> > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> >
> > Bug: webrtc:10358
> > Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27853}
>
> Bug: webrtc:10358
> Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27896}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ied6d9fb96aafe9c957f2658b34b5331b1f359b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135986
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27917}
