This flag is no longer used in Chrome and can now be removed.
Bug: chromium:1314868
Change-Id: Id91b3352dc7ec0543d54894cc206a6e0c7667e9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309960
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40337}
This annotates all unannotated members in VideoCaptureImpl and its
subclasses with either of:
- api_checker_: access on the api thread only
- capture_checker_: access in callbacks/on the capture thread while
capture is active, on the api thread otherwise
- a Mutex if it is already protected by it
Bug: webrtc:15181
Change-Id: I5084e7752a4716c29b85a9b403a88696f66d811f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305647
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40335}
This CL allows the users to now enable/disable WGC capturing support
for Window and Screen sharing independently.
Bug: chromium:1314868
Change-Id: Ieeb15539434dac2caf29c515aa7c5dbb7abcc5df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309560
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40330}
This change fixes a minor issue where we previosuly assumed that the
following was true:
RTC_DCHECK_EQ(map_info.RowPitch, current_frame->stride())
It turns out that this is not always the case when sharing a window
where the stride can sometimes be a few bytes smaller than the
rowpitch.
The code is behind a command-line flag and no tests are affected.
Given limited review resources I therefore plan to bypass the CQ.
I know that it is not recommended but the change has been tested
locally on two different Windows platforms and it does avoid an
existing crash.
Code-Review: alcooper@chromium.org
No-Try: true
Bug: chromium:1421242
Change-Id: I01e7105a6f9fca7ce1349a57635dd373c28d160b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309342
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40308}
This CL adds a light-weight detection of "frame content has changed
since last frame" to an existing pass where bytes are copied from a
texture to a DesktopFrame. The resulting boolean can then later be
used to bypass a full detection of if the content is static or not.
As a result, we only check for static content for a small fraction of
all captured WGC frames and this reduces the total load when 0Hz
is enabled for WGC.
Both WGC and 0Hz support for WGC is still behind a flag.
Bug: chromium:1421242
Change-Id: If9e3721c60a244a3919758fe861d56d4b54cb039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308821
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40299}
VideoCaptureV4L2 has some members that are set in StartCapture during
configuration of the device, but later accessed both on the capture
thread and on the api thread (same as StartCapture) in
CaptureSettings(), which can be called at any time. This is safe because
they are written only on the api thread when the capture thread is not
running. However, there is no helper class that separates the read and
write modes to allow annotations and static analysis of the thread
access of these members.
This commit allows annotations to be added by making VideoCaptureV4L2
use:
- VideoCaptureImpl::_requestedCapability for storing those members
requested through StartCapture(), to allow access on the api thread
through CaptureSettings().
- A new member configured_capability_ to replace those members mentioned
in the first paragraph above. Writes to it happen only in
StartCapture() and StopCapture(), while reads happen exclusively on
the capture thread in between.
Bug: webrtc:15181
Change-Id: I27e0f578e6ac2a2e6b0e34fbabfe4f743b296321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306122
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40290}
CaptureSettings() can read started_ on the api thread at any time. But
it is written and read in pipewire callbacks, on other threads. A lock
seems suitable.
Bug: webrtc:15181
Change-Id: I3d26f02408a4e56921b955f059e504ffa9b8cae9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306121
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40286}
frameInfo_ is used on multiple threads. This commit splits it into:
- VideoCaptureImpl::_requestedCapability for writing what was requested
in StartCapture() and for reading in CaptureSettings(), on the api
thread only.
- A new member configured_capability_ (renamed from frameInfo_) for
accesses in callbacks, or on the api thread when no callbacks can
happen.
Bug: webrtc:15181
Change-Id: I105d8adfde52320e43ffe95fe23e11d028c80684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306120
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40285}
A comment says SetApplyRotation could deadlock if grabbing the lock, but
it does not make any calls under the lock so that is impossible, unless
the caller of SetApplyRotation involves a second lock that the callback
tries to grab, in which case it appears to be a problem of the caller.
Bug: webrtc:15181
Change-Id: Ie16cb01ffb84e9118dd5c87863c29bd107a6c94e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305646
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40279}
Change implementation of `FinalizeFecHeader` to write the FEC header
for multiple ssrcs according to the updated RFC.
Change-Id: I280964b2e53c3579f348fbd42815c966840375ac
Bug: webrtc:15002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307601
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40270}
Changed FinalizeFecHeader to recieve a list of `ProtectedStream` struct,
in order to prepare for receiving multiple ssrcs to protect in the same
FEC packet header. Implementation of the multistream case will follow in
next CL.
Change-Id: I697ef9172a07797a6f500b9ec3a9916f8f45bc04
Bug: webrtc:15002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307620
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40269}
VoipCore still use RtpSenderEgress::NonPacedPacketSender, therefore
packets sent using NonPacedPacketSender::EnqueuePackets are proxied
to the worker thead.
When NonPacedPacketSender is used, the Pacer already guarantee that packets are sent on the worker queue.
Lock is removed from RtpSenderEgress and instead calls must be made on
the worker thread.
Bug: webrtc:15209
Change-Id: Iaf03377ad8a037ecedbbe588a4c1e8e4eadacd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306960
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40252}
Allow absolute send time to go back in time as long as there has not been a large gap in arival time. Use the first packets arival time as time base.
Bug: b/282153758, webrtc:15230
Change-Id: I8663079ab9c202079bf8db303353918d46ba1d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40251}
Header metadata such as audio level and capture time doesn't make sense
for redundant payloads (i.e. RED and inband-FEC).
It is assumed that one of the parsed payload timestamps will correspond
to the RTP header timestamp.
This fixes a bug where capture time and CSRCs were not set after
parsing RED packets.
CreateRedPayload test function is adapted from red_payload_splitter_unittest.cc
Bug: webrtc:15185
Change-Id: Iba58772499b6d760f516854999b60511896b053c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305700
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40240}
Use 8 threads for > 720p
Use 4 tile columns and 2 tile rows for 8 threads
Use 2 tile columns and 2 tile rows for 4 threads
For VGA, 2 tile_col x 2 tile_row configuration has
- ~9% speedup over 4 tile_col x 1 tile_row
- ~5% speedup over 1 tile_col x 4 tile_row
Bug: None
Change-Id: I3c1ea948437aece7c6734ce25351158cbdf0a15b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307880
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40237}
As per the comment in https://webrtc-review.googlesource.com/c/src/+/303240
on the flexfec_header_reader_writer2.h, renaming this file to flexfec_header_reader_writer.h
and renaming the current implementation to flexfec_03_header_reader_writer.h
as it is based on the 03 draft of the RFC.
Change-Id: I80cb2aba6225ec7cd989a134c3204d1db0ac6f7c
Bug: webrtc:15002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40231}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Empty codec objects do not make sense. Instead of creating an empty
object to be used as a placeholder in the API, at least create a
video codec with the right name.
Bug: webrtc:15214
Change-Id: I705d9d1361f353fe5dc538a6fe972c8a346f1247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40218}
Most of the usage of the H264Encoder::Create(codec) method passes a
simple codec with just the H264 codec name. This simplified the call
sites in many places and removes references to the codec types.
Bug: webrtc:15214
Change-Id: I4039c0be4ce6e3147c14c7853df4635f344b7d70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307222
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40214}
This makes it possible to add a SequenceChecker guard to _deviceFd that
ensures it is accessed only on the api thread while the capture thread
is not running, and only on the capture thread otherwise.
Bug: webrtc:15181
Change-Id: Ibc414ee973a3c4798e38e9b9a63e3053b95b9599
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305645
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40194}
Currently FecController knows about network conditions, these information can be used to control RTX settings in-call.
Change-Id: I8f84164aeac48ea13b7f1cf82fd7424431f98ada
Bug: webrtc:15167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304800
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40192}
Add an API to pass AudioFrameProcessor as a unique_ptr.
Bug: webrtc:15111
Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40187}
This change changes the flexfec header reader ReadFecHeader function to parse the FEC header according the the updated RFC. The fec_packet argument is expected to have the protected ssrcs list already populated, as they should be retrieved from the RTP header.
Updated and added Reader unittests. Unittests that are relevant for the Writer, were put inside a comment. In the next change set, when the header writer will be updated, we will update the unittests accordingly.
Bug: webrtc:15002
Change-Id: I118303e31c15c356ffeb2c0aafe503cf293bcad6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40172}
We use value -1 on over all the places through our code so it might be
better to define a constant and use it instead to make the code more
understandable on first look.
Bug: webrtc:15203
Change-Id: I4fc3e561bc7a7778c43ec6cfde7acebef2af79e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306620
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40156}
Allows to use camera portal separately in implementations where each
implementation needs to be called in different places.
This is targeted for Firefox support, where we need to ask for camera
access in the FF frontend code, otherwise making camera access requests
in the backend WebRTC code might result into presenting portal dialogs
asking for access from the javascript API.
Bug: webrtc:15202
Change-Id: Ida8b010bb93e08a9e5ddd9dd8a2a3549ee7fde8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305222
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40148}
This is a partial revert of the previously landed CL in
https://webrtc-review.googlesource.com/c/src/+/301200.
I noticed when I used `getDisplayMedia` on my private Windows laptop
in combination with WGC that the captured screen was distorted and
did only contain tilted (~35 degrees) lines in all sorts of colors.
The issue only happened for one particular screen resolution
3000 x 2000 and 200% scale. Changing to 100% scale instead resolved
the issue.
I tried many other resolutions but could only trigger for the one
above with 200% scaling.
Next, I bisected and found [1] which led to [2] which contains my own
change in https://webrtc-review.googlesource.com/c/src/+/301200.
The only part that could affect the video frame was the part which
did `CopyPixelsFrom` so I reverted that part and it solved the issue
on my private Windows laptop.
I did not dig deeper into why this particular resolution triggered
the distortion but deiced to revert to avoid more reports.
[1] b4c2a7fcf5..ff848b7a43
[2] a9a2957dbc
Bug: chromium:1428592
Change-Id: I328e77840cd3ca6871254cdf06500bdc616b0c36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306600
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40147}
Instead of using most recent, or most "valuable" packets for padding, use most recent large packet.
The large packet for padding is not culled when acked by the receiver.
The most recent large packet is kept where payload size + 100bytes > currently stored.
Bug: webrtc:15201
Change-Id: I510735b757f99460c477b575061963d2b69016e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306521
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40146}
which is the correct term used in
https://www.rfc-editor.org/rfc/rfc3611#section-4.4
BUG=None
Change-Id: Iab5a1de6b69a8495aa9a6f79531053f4f2421c27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306480
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40143}
Replace Transport* interface with since std::function to stress this class doesn't produce RTP packets
Repesent outgoing packet as ArrayView instead of pointer + length.
Make outgoing transport optional, thus allowing to use RtcpTransciever as an rtcp parser.
Bug: webrtc:8239, webrtc:14870
Change-Id: Ia582d9a980786df8e295adcebe27081258b80dc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306280
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40134}
This reverts commit 8856410b6d54b546bdb3185587474f0f9b3a7c2e.
Reason for revert: chromium:1447540
Original change's description:
> pipewire capturer: Reduce the amount of copying
>
> Improves the capture latency by reducing the amount of
> copying needed from the frame. We keep track of the
> damaged region of previous frame and union it with
> the damaged region of this frame and only copy this
> union of the frame over. X11 capturer already has
> such synchronization in place.
>
> The change is beneficial especially when there are
> small changes on the screen (e.g. clock ticking).
> For a 4k screen with 128 cores, I observed the
> capture latencies drop from 5 - 8 ms to 0 ms when the
> system is left idle. This is in line with the X11
> capturer.
>
> Bug: chromium:1291247
> Change-Id: Iffb441f9e1902d2658031f5f35b5372ee8e94073
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299720
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Salman Malik <salmanmalik@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#39968}
Bug: chromium:1291247
Change-Id: Id1bfd3fc39fea2bb1f232cad5218f90e144920e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306263
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40123}
