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}
Update some uses of scalabilityMode to also set scaleResolutionDownBy.
Nothing changes in these tests because they were either single stream
cases, fallback was happening anyway or the test was doing VP8, but by
specifying scaleResolutionDownBy we make these tests consistent and
explicit about wanting to exercise the "standard path".
Partly a style thing and partly not wanting the test to pass for the
wrong reason.
Bug: None
Change-Id: I1d2e688976a4e6c160e90474e2416a18d795d41b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299078
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39685}
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}
FieldTrialBasedConfig reads config from the global field trial string
ScopedKeyValueConfig adjust the global field trial string
test::ExplicitKeyValueConfig doesnt touch the global field trial string
Bug: webrtc:11926
Change-Id: I8883634fdc7e1bdb63eec9bf38114a3031103839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299062
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39683}
This updates DataChannelController and test classes to use
GetSctpSslRole_n instead and query the role on the network thread.
Along the way this CL makes the init config struct for when constructing
data channels, mandatory. It's now passed via const& instead of by pointer. In practice a valid pointer was always being passed.
Bug: webrtc:11547
Change-Id: I0f4bbf364969cc2dec07871c297ddbef0c175f86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298307
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39676}
Replace locally-defined keys and values with constants from MediaCodec API (MediaFormat.KEY_..., etc). Value of a constant field is resolved at compile time according to 13.1.1 [1]. I.e., it is safe to reference a constant field not available in older API (MediaCodec API ignores unrecognized MediaFormat.KEY_ values).
[1] https://docs.oracle.com/javase/specs/jls/se20/html/jls-13.html#jls-13.1
Bug: none
Change-Id: I3c63cfd67cc22db1b957f908508b434d36d88806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298940
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39668}
This is to be more robust to packet loss during DTX and paused streams.
Without it, we can wait to decode an available packet when in CNG or
PLC mode until more packets arrive, which for DTX and paused streams
can take a long time.
We already include the waiting time if the last packet in the buffer
is a DTX packet.
Bug: webrtc:13322
Change-Id: Iaf5b3894500140d6f83377ba2cd65b44e0cdac05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299009
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39667}
