And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.
Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
It was previously logged in Call, but streams are not always created
with the full configuration, which caused header extensions to be
missing from the log.
Bug: webrtc:9885
Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131
Reviewed-on: https://webrtc-review.googlesource.com/c/108601
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25483}
The replacements are absl::EqualsIgnoreCase and
absl::StartsWithIgnoreCase. Also delete the alias
RtpUtility::StringCompare.
Bug: webrtc:6424
Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f
Reviewed-on: https://webrtc-review.googlesource.com/c/109006
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25481}
Support Injectable Audio Codecs from the Java SDK.
The PeerConnectionFactory.Builder defaults to
BuiltinAudio(Encoder|Decoder)Factory, but other implementations are
permitted via the Audio(Encoder|Decoder)FactoryFactory interface.
Bug: webrtc:9916
Change-Id: I61ad4a6e57666bc1be79daf5f40b129e0eacad84
Reviewed-on: https://webrtc-review.googlesource.com/c/107711
Commit-Queue: Lennart Kolmodin <kolmodin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25478}
This CL refactors AGC2 and fixes the order with which the fixed
and the adaptive digital gain controllers are applied - i.e., fixed
first, then adaptive and finally limiter.
FixedGainController has been removed since we need to split the
processing done by the gain applier and the limiter.
Also, GainApplier and Limiter are easy enough to be used without
a wrapper and a wrapper would need 2 separated calls in the right
order - i.e., error prone.
FrameCombiner in audio mixer has been adapted and now only uses the
limiter (which is what is needed since no gain is applied).
The unit tests for FixedGainController have been moved to
gain_controller2_unittests. They have been re-adapted and
ChangeFixedGainShouldBeFastAndTimeInvariant has been re-tuned.
Bug: webrtc:7494
Change-Id: I4d7daeae917257ac019a645b74deba6642f77322
Reviewed-on: https://webrtc-review.googlesource.com/c/108624
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25477}
GetInputPin() and GetOutputPin() do not guarantee returning non-null pins.
Should either of them return null during Init() we're better off returning an
error than allowing unsafe behavior further ahead.
Bug: webrtc:9941
Change-Id: I25858f0555334b4ef99801f83454931384695bf6
Reviewed-on: https://webrtc-review.googlesource.com/c/108603
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25475}
IceConnected state (transport state) now includes the state of the
MediaTransport.
This is a first change of two. Second change will add state change
signals to the PeerConnectionInterface informing separately about
ice+media transport vs ice+dtls.
Bug: webrtc:9719
Change-Id: I5731530073e8f26dfc8b188778d268b815da7052
Reviewed-on: https://webrtc-review.googlesource.com/c/108901
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25473}
The extra saturation margin is a setting for the SaturationProtector
in GainController2. The higher it is, the less gain GC2 will apply. In
this CL we pipe the setting up to audio_processing.h. Now the setting
can be set at a high level.
Also in this CL add a few (missing, they should have been there
already) tests for the GC2 and GC2 with saturation margin.
Bug: webrtc:7494
Change-Id: I1b61f1662e6c6a8817fd5b0e845339694bf8d50d
Reviewed-on: https://webrtc-review.googlesource.com/c/109001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25470}
The frame time deltas are now capped based on the current noise.
This has been tested in various conditions using both screen content
and typical mobile video settings, to produce delays that are not overly
high screen content, and simultaneously not negatively affect mobile
calls on really bad network that may have high natural jitter.
Bug: webrtc:9898
Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48
Reviewed-on: https://webrtc-review.googlesource.com/c/107349
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25469}
This is a preparation for deleting rtc::PacketTime. Next step, after
downstream code has been updated to not access the |timestamp| member,
is to make rtc::PacketTime an alias for int64_t.
Also delete the unused member rtc::PacketTime::not_before.
Bug: webrtc:9584
Change-Id: Iba9d2d55047d69565ad62b1beb525591fd432ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/108860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25468}
--force_fieldtrial=/WebRTC-DecoderDataDumpDirectory/./ will create a
file ./webrtc_receive_stream_[ssrc].ivf containing the exact data that
is fed to the decoder.
Bug: None
Change-Id: I4042298c9b851fc4b61c417652315fa2610de1ed
Reviewed-on: https://webrtc-review.googlesource.com/c/107644
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25467}
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.
To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.
Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
This CL prevents dereferencing potentially null pointer by
setting the pointer in client code.
We can now safely call PeerConnection::Close(), which happens
to trigger OnIceConnectionChange() on the observer.
This is a followup to: https://webrtc-review.googlesource.com/c/src/+/107706
Bug: webrtc:9855
Change-Id: Ieebf8415f0a12fe87d8cd80d1eb06797926005df
Reviewed-on: https://webrtc-review.googlesource.com/c/108785
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25462}
Currently, when users want to use the screen sharing and are using the
Wayland display server (the default on Fedora distribution), then it
doesn't work, because the WebRTC only includes the X11 implementation.
This change adds the support by using the PipeWire multimedia server.
The PipeWire implementation in WebRTC stays in
screen-capturer-pipewire.c and is guarded by the rtc_use_pipewire build
flag that is automatically enabled on Linux.
More information are included in the relevant commit messages.
Tested on the current Chromium master and Firefox.
The sysroot changes are requested in:
https://chromium-review.googlesource.com/c/chromium/src/+/1258174
Co-authored-by: Jan Grulich <grulja@gmail.com>
Co-authored-by: Eike Rathke <erathke@redhat.com>
Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
BUG=chromium:682122
Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
Reviewed-on: https://webrtc-review.googlesource.com/c/103504
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25461}
This drops the locks and annotations in EchoCancellationImpl,
now that the interface is no longer externally accessible.
Bug: webrtc:9929
Change-Id: I401256f523340cbabce23a5914ab28ce44179935
Reviewed-on: https://webrtc-review.googlesource.com/c/108602
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25460}
to Mdns.*.
MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.
MDns.* is also renamed to Mdns.* per the style guide.
TBR=aleloi@webrtc.org
Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
This change deletes the default implementations of state and data
channel methods (SetMediaTransportStateCallback, SendData, CloseChannel,
and SetDataSink). It adds stub implementations to LoopbackMediaTransport
and FakeMediaTransport.
Bug: webrtc:9719
Change-Id: I49b7780c055b552330546b460c2e79ce8df81833
Reviewed-on: https://webrtc-review.googlesource.com/c/108940
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25457}
Adds the types and methods required for sending and receiving data
channel messages over the media transport. These are:
- A DataMessageType to distinguish between text, binary, and control
messages
- A parameters struct for sending data messages, which specifies the
channel id, type, and ordering/reliability parameters
- A sink for data-channel related callbacks (receive data, begin
closing procedure, and end closing procedure)
- A method to set the sink for data channels
- Methods to open, close, and send on data channels
These methods, combined with the state sink, allow PeerConnection to
implement the DataChannelProviderInterface using MediaTransport as the
underlying transport.
Change-Id: Iccb2ba374594762a5b4f995564e2a1ff7d8805f5
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/108541
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25454}
This allows to use secure, end to end communication if SDES cryptos are
passed. MediaTransport can use a derived key to secure its own
communication.
Bug: webrtc:9719
Change-Id: If1a20b136b3b4af0cb24f10b52fc5ce1eb31daa2
Reviewed-on: https://webrtc-review.googlesource.com/c/108504
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25452}
This makes the calculation more similar to the one in WebRTCVoiceEngine.
Bug: webrtc:9510
Change-Id: Ibca69842726e51c07b9cc9550ff9f15a24161e28
Reviewed-on: https://webrtc-review.googlesource.com/c/107653
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25448}
Moved methods: GetReadData, ConsumeReadData, GetWriteBuffer,
ConsumeWriteBuffer, GetWriteRemaining.
These methods represented an optional interface for reading and
writing streams, intended to optimize certain use cases. However,
it was implemented only in the FifoBuffer subclass, and the few
users of that class all have a concrete FifoBuffer, and hence
don't need the methods on the abstract StreamInterface.
Bug: webrtc:6424
Change-Id: I6de74d1a9205fcb7037ad84e24679d4a27c1d219
Reviewed-on: https://webrtc-review.googlesource.com/c/108621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25446}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
When FlexFEC is enabled, sometimes media packet will be recovered by FEC before the actual media packet's arrival. In current implementation this will be considered as packet out of order and nack will be sent, thus cause large increase in retransmit bitrate.
This fix:
1. Avoid sending nack for packet out of order caused by "early" recovered media packets.
2. Save recovered media packet in a set, and do not send nack for these packets.
Bug: None
Change-Id: I008ef4e33668bce6d2cb9ff52b4b5c8e3f349965
Reviewed-on: https://webrtc-review.googlesource.com/c/108090
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25444}
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.
On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.
Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
This change just wraps the openssl key derivation functions in a simple
interface in a similar way to how we do it for messagedigest.h so we aren't
coupled to openssl in the core implementation.
Bug: webrtc:9917
Change-Id: I8556bd6e38b7da34d93abbe29415c3366f6532ba
Reviewed-on: https://webrtc-review.googlesource.com/c/107981
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25440}
This class exposes Wait()-Set() logic to synchronize events.
- There is a bug in checking EventWrapper::Wait() as it returns [1,2]. Negating
these values cause us to always pass timeout checks.
- There is a general problem in this class with waiter. There are 2 scenarios:
1) Lock()-Unlock()-DisplaysReconfigured()
In this scenario, Wait() in DisplaysReconfigured() immediately passes as event
is already signaled. Next Lock() call won't continue until Set() is called in
DisplaysReconfigured(). This blocks capture thread from accessing display until
reconfiguration completes.
2) Lock()-DisplaysReconfigured()-Unlock()
In this scenario, Wait() in DisplaysReconfigured() passes when Unlock() called.
Capture thread accesses display while reconfiguration happens. Note that we are
only delaying the OS delegate thread here. As an experiment, adding Sleep() in
DisplaysReconfigured() results in no change, because it looks like OS uses this
thread for only delegates but not for the actual display switch.
Overall, (1) doesnt seem necessary as (2) already accesses display while
reconfiguration happens. (2) doesn't seem necessary as blocking system delegate
thread doesn't help. Therefore, I changed the class to only protect from race
condition on |desktop_configuration_|.
Bug: chromium:796889
Change-Id: I37263305e5ac629e21ff9e977952cf4a21bae19f
Reviewed-on: https://webrtc-review.googlesource.com/c/108560
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25437}
