This fires in downstream tests when re-negotiation of a datagram
transport completes.
Bug: webrtc:9719
Change-Id: Ie7337e7dc33e41a83da37d3ef2fda596d9107256
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140821
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28183}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28182}
As this is handled higher up the pipeline in a single
place for all encoders/decoders
Bug: webrtc:10460
Change-Id: I95b0a69aecaf07283c8776ac0d7e85d097e3576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139882
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28172}
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}
Using this flag, an encoder may inform the RTP sender module that
the packet is not elligible for retransmission. Specifically, it
may not be retransmitted in response to a NACK message,
nor because of early loss detection (see CL #135881).
Bug: webrtc:10702
Change-Id: Ib6a9cc361cf10ea7214cf672e05940c27899a6be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140105
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28169}
Some of the TODOs associated with webrtc:10336 which are
currently in the codebase have recently been resolved,
but not all relevant TODOs have been removed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10336
Change-Id: Iff1d0fc94dee5bf49226f6ea3d9127fea77e9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139902
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28167}
Replaced by separate methods
SendPictureLossIndication and SendFullIntraRequest.
The split SetKeyFrameRequestMethod/RequestKeyFrame implicitly
requires that the two methods are called on the same thread, to avoid a
data race. After downstream code is updated, both deprecated
methods and the member |ModuleRtpRtcpImpl::key_frame_req_method_| can
be deleted.
Bug: None
Change-Id: I454f6d16b667f2306cba0dec467ddc183ad449c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140043
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28163}
Reset() is called each time the encoder is reconfigured, but then it
happens the target bitrate isn't reset in encoder. So it might produce a
frame before next bitrate estimate is propagated to the metadata writer.
The incorrect zero bitrate would be treated as a paused encoder and would
cause metadata to be dropped.
Also, added unittest for that scenario at VideoStreamEncoder level.
Bug: webrtc:10460
Change-Id: I28024a527f1fb8474b172e2c5c2394fd38d69a07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140101
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28159}
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}
SurfaceTextureHelper currently crashes if an OES texture is produced
before setTextureSize() has been called. This is annoying if the texture
size is not easily known beforehand. A real world example is MediaPlayer
that provides the video size with an asynchronous call to
setOnVideoSizeChangedListener(), but that might happen after the first
texture is produced on some devices.
This CL waits with delivering frames until the size has been sent,
rather than crashing.
Bug: webrtc:10709
Change-Id: I5d9ce542e0edaafe1153fd5fe7d64dba86d7e33c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140080
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28151}
- This makes it consistent with ICE and MediaTransport ownership.
- Removes unnecessary datagram_transport() getter in DtlsTransportInternal
As a side effect this fixes bug in JsepTransportController, which moved datagram_transport to Dtls after creating it, then checked if (datagram_transport) to decide which RTP transport to create. As a result of this bug we were creating Sded instead of Unencrypted RTP with datagram transport.
Bug: webrtc:9719
Change-Id: Ic5b13a450ce6ac5b2a20d388657e3949aabef079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139620
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@{#28146}
This documents in the API what is already true in the
implementation - that SessionDescription will eventually
delete MediaDescription objects passed to it.
The old API is preserved for backwards compatibility, but
marked as RTC_DEPRECATED.
Bug: webrtc:10701
Change-Id: I9a822b20cf3e58c5945fa51dbf6082960a332de8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28144}
This is an usual error while using neteq_quality_test. This tool
does not support wav files as input. Adding a validation.
Bug: webrtc:10690
Change-Id: I18ed308d2f688106728df5df25e0a58c7170f411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139104
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28141}
The test RunPlayoutAndRecordingInFullDuplex makes the speakers play the
signal it simultaneously records from the microphone, which can cause
full howling.
The test itself measures buffer usage and does not depend on what signal
is played through the speakers. This change mutes the speakers to prevent
howling when running modules_unittests.
Bug: webrtc:10704
Change-Id: I6176adb2fb5ce0e4d86f6f447476be8a88c2f2cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139889
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28140}
This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet.
Bug: webrtc:10668
Change-Id: I962df493a76692f668314f78d6792d7636c5a31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28138}
Removes CAPTURE_VIDEO_OUTPUT permission since it is not needed
when using MediaProjection API and not allowed in Android Q.
This is a reland of af4f1b41277ebdf0d7386cbd2903abc709cbc183
Original change's description:
> Target SDK level 27 in AppRTCMobile.
>
> Implements the dynamic permission model required by the newer SDK and
> changes the theme.
>
> Bug: webrtc:8803
> Change-Id: I3ea23a25b27f196fcffd018c7cdd2ff6255b62d9
> Reviewed-on: https://webrtc-review.googlesource.com/44400
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21788}
Bug: webrtc:8803
Change-Id: Ie79d7f7b9e8fea255581d3b56772b5bb6fffff4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139881
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28137}
Currently, if LNTF and NACK messages are both created, they will
be sent out in separate RTCP messages. This is wasteful.
This CL is the first of in a series of CLs that will ensure that
these feedback messages can be buffered together, without introducing
more of a delay than the CPU time required to process both messages.
Bug: webrtc:10336
Change-Id: I950324112ee346695a12a17d025483ea5e99c732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139112
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28136}
This change makes SwapQueue lockless in order to reduce lock contention
in the Audio Processing Module.
Bug: webrtc:10205
Change-Id: Idc3b2a85e959b467bc1653492e48eee42e236fa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138901
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28135}
