(using no-try due to bot infra issue)
No-try: true
Bug: b/276434297
Change-Id: I33f796b501f96731c4ca76cb62c2331f10c795f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299708
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39734}
This is needed in order to be able to update the legacy stats
collector to fetch data channel stats from the network thread, which
is part of an upcoming change to data channels.
Bug: webrtc:11547
Change-Id: Ic205b0314b9f11a024d36d714c223cbddd0f3df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299462
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39732}
The goal is to reduce the amount of time stretching done in response
to network jitter. Specifically, we should be able to “ride” over delay
spikes if the current delay is sufficient, without decelerating
playout. We should also avoid accelerating immediately after a buffer
underrun, until we are reasonably sure that the jitter has passed.
This is achieved by increasing the deadband where we choose to do
normal playout, based on the maximum delay in the short term packet
arrival history.
The buffer level filter is still used to report the average delay for
A/V sync purposes.
The new behavior is behind a flag and will be experimented with before
it is made default.
Bug: webrtc:13322
Change-Id: I5fba0c9d46d835dbe5401669598fa031512ccced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39730}
With this cl, packets that are discarded in RtpTransport now notifies Call, so that
they can be part of BWE even if they are dropped.
These packets have been recevied on the transport, and has bin decrypted
and parsed and thus can be accounted for.
The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded.
Bug: webrtc:14928
Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39727}
in order to not regress existing use-cases while following rules
described by the specification. This change now makes the existing
regression test pass after the spec-compliant modifications.
BUG=chromium:1051821
Change-Id: Ia384adf9a172ed88b5ec6a3cc5c478764a686cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39726}
The tests now have two ways of addressing the channel under test:
* channel_: All SctpDataChannelTest methods. This way calls get
executed as specified by the proxy and more accurately reflects
production.
* inner_channel: For SctpDataChannel methods that are not a part of
the SctpDataChannelTest interface. Use this to invoke event handlers
etc.
Upcoming changes will include threading changes and changes to the
proxy, so it's important to cover both.
Bug: webrtc:11547
Change-Id: I26c284ece82b9a58e2b5dc4468d124d54012d959
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299264
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39725}
* Rename id_ -> id_s_, add id_n_ and thread guards.
* Same for getters, sid() -> sid_s(), add sid_n()
As more things migrate over to the network thread, we'll only need the
_n variant.
Bug: webrtc:11547
Change-Id: Ic998330f4c81b0f6833967631ac70edc2ca2301c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299141
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39724}
It is a part of "encoding statistics" feature [1] available in Android SDK 33. Local testing revealed that for HW VP8/9 encoders we get QP in range [0,64] which is not what WebRTC quality scaler expects. Exclude VP8/9 encoders for now.
[1] https://developer.android.com/reference/android/media/MediaFormat#VIDEO_ENCODING_STATISTICS_LEVEL_1
Bug: webrtc:15015
Change-Id: I8af2fd96afb34e18cb3e2cc3562b10149324c16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298306
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39722}
The reason is that:
* SDP/PeerConnection code should use HasUsedDataChannels() instead.
* HasDataChannels() touches the list of data channels on the signaling
thread, which will be a problem when it's moved to the network
thread.
* It's only needed by tests.
Bug: webrtc:11547
Change-Id: Idd47365c429e5f1d6e3812cf558c4e6fefbf733c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299481
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39721}
...when checking if negotiation is needed.
I noticed this call site after we recently updated PeerConnection here:
https://webrtc-review.googlesource.com/c/src/+/297860
Bug: chromium:1423562
Change-Id: Id37c938d731eadfccff44c95ef757a3cabd64936
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299480
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39720}
* Change data channel creation code to return RTCError for more
detailed/accurate errors.
* Move DataChannelController::sid_allocator_ to the network thread.
* Add a temporary duplicate vector of channels on the network thread.
This will eventually be the main vector.
* Delete one test that turns out to be racy (as long as we're using
both the signaling and network threads).
Bug: webrtc:11547, webrtc:12796
Change-Id: I93ab721a09872d075046a907df60e8aee4263371
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298624
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39719}
This is to avoid counting concealed samples after comfort noise as
speech.
Bug: webrtc:13322
Change-Id: I12cf18d720c697d81376c6f6cdc02d7c6bfa49a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299300
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39717}
MediaType::Any will be used by packets that can not be demuxed by
RtpTransport.
Bug: webrtc:14928
Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39710}
Pass FieldTrialsView by const& to note it can't be null and doesn't need to outlive the constructor
In unittests use AimdRateControl object directly instead of through helpers
Use unit types (TimeDelta, DataRate) directly, reducing their conversion to plain numbers
Replace SimulatedClock with a single Timestamp now variable or constant
Bug: None
Change-Id: I147f85e629b4d8923aa19896ea211a6f9dca1e68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299260
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39707}
UpdateLastDecodedPacket is anyway only called when a new packet is
decoded.
Bug: webrtc:10178
Change-Id: I8cfcc5791e71079034a2d0806c44b3b071ac2ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299180
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39703}
This will be included in INIT/INIT_ACK if the socket has
configured the "accept zero checksum" parameter, that will
be added in follow-up CLs.
Bug: webrtc:14997
Change-Id: I1a2823fbc77cfea8fe746b07c1c77593bc15efe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39702}
This flag isn't needed for sctp data channels.
Bug: none
Change-Id: I07b8ba2c5186729b8a5edb4d2bba7b800335ab5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299074
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39701}
which describe the existing behavior that necessitated the revert
b396e2b159b060791954495d68278a55e8f72092
Also change the fake media engine audio clockrate to 8000 instead
of 0 and the fake media engine video payload type to something but
0 as this value seems to be treated specially by the video engine
and is a payload type reserved for PCMU.
BUG=chromium:1051821
Change-Id: Ib0a345d59baba50a565f01685d240e41584367e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299000
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39699}
when there are lot of missing packets to report before next received packet
Bug: chromium:1426582
Change-Id: I6746294152d13e18120cdb173b11b245e5c803f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39698}
The EncoderStreamFactory triggers different code paths depending on
`number_of_streams`: one for simulcast and one for non-simulcast.
The non-simulcast path is desired for both normal streams and SVC
streams.
The simulcast path gives sensible max bitrates for 4:2:1 scenarios, but
when encodings like {active,inactive,inactive} is specified in order to
do standard SVC, the max bps of the first encoding is so low that an
SVC stream will never send more than its first spatial layer (even when
scaleResolutionDownBy is 1).
Because of this, standard SVC is broken. This CL fixes this problem by
using the CreateDefaultVideoStreams() code path instead, which is the
same one that legacy SVC uses. With this fix, legacy and standard SVC
produce the same behavior regarding bitrate.
An added benefit of this is that numberOfSimulcastStreams == 1 in the
standard SVC path as well.
{active,inactive,inactive} tests are updated to verify the full
resolution is achieved after ramp-up. I've also confirmed that this
fixes the bug in Canary, see https://crbug.com/1428098#c2.
Bug: chromium:1428098, webrtc:15041, webrtc:15034
Change-Id: Ia1eb4ff59c4e2a56af833f7ac907a66bca8ea054
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299147
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39697}
* The pointer isn't needed for this notification. Arguably using
the internal id is more consistent with the stats code.
* Using the int makes it safer down the line to post the operation
from the network thread to the signaling thread rather than post
an object reference.
Bug: none
Change-Id: I1e9eb31d8386dca3feaa90ee3267ea98eb3e81ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299144
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39696}
This change makes it more clear that there are three different
capture API combinations for screen sharing on Windows.
No option => GDI without fallback.
DirectX option => DirectX with GDI as fallback
Magnifier option => Magnifier API with GDI as fallback
Previously, if both DirectX and Magnifier were enabled, the end
result was Magnifier for unknown reasons.
With this change in place, we can remove usage of the Magnifier API
in Chrome and switch between DirectX and GDI simply by allowing
DirectX or not.
Bug: None
Change-Id: Ice915d6721fa84a25d275f22246df73fc61f64b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299061
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39692}
Make DataChannelController's AddSctpDataStream and
RemoveSctpDataStream be required to be called on the network thread.
This moves blocking calls within those methods over to the
SctpDataChannel class instead.
For production code there's no functional change in this CL. However, this CL:
1) Introduces an actual dedicated network thread to
DataChannelController and SctpDataChannel tests.
2) Removes two data_channel_transport() checks inside DCC that
were being done on the wrong thread (signaling) and
3) introduces a network calling block to SctpDataChannel, where more
network thread related work needs to be done and can be bundled.
(to be done in follow-up CLs).
Bug: webrtc:11547
Change-Id: I6787ac395e61d4a25ae3a74a123e3357cbb46b54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298052
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39688}
This reverts commit cac9a55ddf0ba33f7407f707b69d66d01c49073b.
Reason for revert: WebRTC importer consistently fails this test.
Original change's description:
> Add legacy SVC test that all layers can be inactivated.
>
> A larger version of this test was previously landed but got reverted
> due to failures only happening on the importer bot (not on the CQ or
> locally).
>
> This is a smaller version of the test that does something we should
> support: being able to inactive all encodings of a VP9 legacy SVC
> stream.
>
> Let's land and see if any issues are reproducible (expecting revert).
>
> Bug: webrtc:15033
> Change-Id: I88da1facf4ef05299f3392b86a0e3df029ebe264
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299006
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39684}
Bug: webrtc:15033
Change-Id: Ia31f125bd6782ed38653c1e5cdcc29a8a0eff874
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299145
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39686}
