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}
There is still an TSAN issues, but they are inside
usrsctp library, that's why tests are still disabled
by default.
Bug: None
Change-Id: I55f7c66b4d9a5feccd2121e2dd3b131cf1564804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158522
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29640}
Refactorings to the audio processing module has, piece by piece,
decreased the workload of the apm_helpers helpers. It has come to a
point where it seems more reliable to consolidate what little is left
into the WebRtcVoiceEngine itself.
Bug: webrtc:9878
Change-Id: I6d983ace8e7ccb1b99d95178cf72608a657c7506
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157443
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29553}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
Video and audio senders are missing mid, rid and rrid extensions in
their GetCapabilities call.
Bug: chromium:1007894
Change-Id: Ie9edba28ae32fda5e501913cac694f43bfb185ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156560
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29493}
The experiment was extended to support per-codec minimum bitrates
for the following codecs:
* VP8
* VP9
* H.264
The old semantic meaning for the field trial is retained, in that
specifying "br:" applies a minimum bitrate to all codecs. If "br:"
is not specified, the per-codec minimum config is consulted.
Bug: webrtc:11024
Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29450}
Merge GlobalLock and GlobalLockPod, make member private.
annotate creation of all GlobalLocks with ABSL_CONST_INIT
Bug: None
Change-Id: I29abcc86796ec0e45b15df7d26392309d1bf7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29447}
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}
This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.
It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.
Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
Bug: None
Change-Id: I975bc8779bac9700854de411301415338dcaf9f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154820
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29343}
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}
In this CL:
- Renamed EncoderFailureCallback to EncoderSwitchRequestCallback. An encoder
switch request can now also be made with a configuration that specifies which
codec/implementation to switch to.
- Added "WebRTC-NetworkCondition-EncoderSwitch" field trial that specifies
switching conditions and desired codec to switch to.
- Added checks to trigger the switch based on these conditions.
Bug: webrtc:10795
Change-Id: I9d3a9a39a7c4827915a40bdceed10b581d70b90a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29196}
* Increase 540p bitrate to 1.2mbps from 0.9mpbs.
960x540 bitrate was by far smallest in terms of bits per pixel. This change
brings it closer to other resolutions.
* Interpolate max/target/min bitrates for non-standard resolutions based
on number of pixels.
Bug: webrtc:10965
Change-Id: If0aa56bb4c614ca09ee39d3a2b700aab2ffa1a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152828
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29195}
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}
This is a reland of 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9
Original change's description:
> New build target p2p:stun_types
>
> The media:rtc_media_base target needs definitions of various
> stun-related types and constant. With this new smaller target, it no
> longer needs to depend on all of p2p.
>
> Bug: webrtc:8733
> Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29036}
Tbr: steveanton@webrtc.org
Bug: webrtc:8733
Change-Id: I1847007ecf29e0e6a27f559b92df632a1cd69280
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151880
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29092}
This reverts commit 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9.
Reason for revert: Breaks build
Original change's description:
> New build target p2p:stun_types
>
> The media:rtc_media_base target needs definitions of various
> stun-related types and constant. With this new smaller target, it no
> longer needs to depend on all of p2p.
>
> Bug: webrtc:8733
> Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29036}
TBR=steveanton@webrtc.org,mbonadei@webrtc.org,nisse@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8733
Change-Id: I6e00657a6137ff773325f37ec02ee1014b6fe96b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151740
Reviewed-by: Hannes Landeholm <hnsl@webrtc.org>
Commit-Queue: Hannes Landeholm <hnsl@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29085}
Currently we set the EOR bit when sending a message through the sctp
library. This makes the send non atomic, meaning that message can be
partially accepted by the sctp socket. Currently we ignore the sent
amount result, but this change now checks that result and buffers the
remaining message to be sent later in the case that it was only
partially accepted by usrsctp.
Bug: webrtc:10922
Change-Id: I9ff563c40e2b7dbdeb19b40d07c43a15ff7c9b49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150562
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29051}
"WebRTC-VP8-Forced-Fallback-Encoder-v2" affect VP8 only, "WebRTC-Video-MinVideoBitrate" apply to all codec. When both field trial string are set, the bitrate set by "WebRTC-VP8-Forced-Fallback-Encoder-v2" will be used.
Bug: webrtc:10915
Change-Id: I63da5909c04ecfad99e93a535fbf71293890fd11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151135
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29047}
The media:rtc_media_base target needs definitions of various
stun-related types and constant. With this new smaller target, it no
longer needs to depend on all of p2p.
Bug: webrtc:8733
Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29036}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
Also annotate a few of the remaining uses, to guide further splits of
that large build target.
Bug: webrtc:8733
Change-Id: I16ac33ab48e6d39a1a8dbc2a3fc671d8db6dbfe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150789
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29001}
