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}
StreamDataCounters is used both for send-side and receive side stats,
but last_packet_received_time is only used by receive statistician where
it duplicates another member
Bug: webrtc:13757
Change-Id: Iae6a65aba497e577ee3255e40623362e8c4c8a72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306183
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40119}
Create a copy of flexfec_header_reader_writer for changing the implementation according to updated RFC. The fork is needed, since the updated RFC is incompatible with flexfec-03.
In the updated RFC, we receive the list and the number of protected ssrcs from the RTP header (from it's CSRCs , and CSRC count fields).
This Change is only a copy of the existing files. This will make it easier to understand the changes to the implementation in the next change sets.
Bug: webrtc:15002
Change-Id: I31bf5eca0d8f3cb23b4caabb477897eeb0ca6d96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303240
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@{#40103}
which is the maximum allowed in RFC 3550:
The last octet of the padding contains a count of how
many padding octets should be ignored, including itself
SRTP encryption does not need to be taken into account since none of
the cipher suites used by WebRTC require padding:
https://www.rfc-editor.org/rfc/rfc3711#section-3.1https://www.rfc-editor.org/rfc/rfc7714#section-7.2
BUG=webrtc:15182
Change-Id: Ife3d264af389509733699f2dd4d32ba63793e9de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305642
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40101}
Error resilience is no longer required for upper temporal layers.
Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.
Reland of https://webrtc-review.googlesource.com/c/src/+/302001
Bug: webrtc:15106
Change-Id: I72ca9d504a7848dda934cbd52669027061742256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305782
Reviewed-by: Jerome Jiang <jianj@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/main@{#40099}
This change replaces ReceivedFecPacket FEC header fields with vectors (for protected ssrcs, sequence numbers and masks), which is needed to support protection of multiple ssrcs in the same FEC packet (as part of the flexfec RFC - https://datatracker.ietf.org/doc/html/rfc8627).
Bug: webrtc:15002
Change-Id: I82c54203fcfec10c760f34f805cc6308562e3df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303200
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40075}
The IvfFileWriter logs a warning in case frames have a different
resolution compared to the one of the first frame in the file.
While this is an issue, since the IVF header will have the resolution
of the first frame, in reality this is not a problem (e.g. tools like
VLC can open and play the IVF without issues).
For this reason, let's remove the log which gets printed for each
frame.
Bug: b/282678729
Change-Id: I540cd1b6ce4f5d888737725e7615918aa126647f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305280
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40069}
The fcntl() call has variable arguments, therefore we need to pass 0 to
specify there are no other arguments for this call, otherwise we might
end up with an argument that is random garbage.
Bug: webrtc:15174
Change-Id: I34f16a942d80913b667d8ade7eed557b0233be01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305120
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40060}
RTX padding packets sent before media packets can legitimately have no
timestamps set (they are 0). Writing the TransmissionOffset extension
with capture time 0 will overflow once current time exceeds ~3 minutes.
Bug: webrtc:15172
Change-Id: I4dd1f341802d45016549b330f0e08cd3a00cfa19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305020
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40055}
With intent to fully replace RtcpBandwidthObserver interface
and half of the TransportFeedbackObserver interface
RtcpBandwidthObserver interfaces passed bitrate and time variables as
raw ints, NetworkLinkRtcpObserver uses more expressive types.
Bug: webrtc:13757, webrtc:8239
Change-Id: I0a8c8de626fbe0c190a0a1a9f6733d863494401c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304700
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40043}
The VCMReceiveStatisticsCallback interface is both implemented (by ReceiveStatisticsProxy) and called (by VideoStreamBufferController) in `video/`, so there's no reason it should be declared in `modules/video_coding`. I also took the opportunity to update the name.
No functional changes are intended by this change, but following CLs will make some changes.
Bug: webrtc:15085
Change-Id: Ib8da30ca56675e4f638d0b9778c329b9c1138acf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304662
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40034}
RtpRtcpInterface::RTT follows discouraged style of using return values,
uses raw integers to represent time delta,
and returns values that no code uses (min, max, average RTT)
added LastRtt function addresses all these stylistic issues.
Bug: webrtc:13757
Change-Id: Iaf947dd1b7139026f2beb991e69634c606c6b608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304520
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40028}
Same information is already passed using ReporBlockData class
Bug: webrtc:8239
Change-Id: Iaae0edd34941c45527414a20923b5238e7a822fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304641
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40021}
This CL removes a PostTask in response to packet receipt reception.
This is made possible due to PacketRouter lock removal in
https://webrtc-review.googlesource.com/c/src/+/300964.
Depending on how transport code is organized, this may lead to
possibility of packet receipts arriving in
RtpTransportControllerSend which may re-enter the PacingController's
ProcessPackets method, leading to out-of-order packet sends. Fix
this by detecting re-entry and avoiding a second ProcessPackets call
in the TaskQueuePacedSender.
Bug: chromium:1373439
Change-Id: I24928f2d28a240d0860fe7e4a114cedf1f13d2bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40017}
Remove deprecated accessors returning time as raw int
Add setters for all fields to simplify usage of this class in tests
Remove unused min/max RTT fields
Bug: webrtc:13757
Change-Id: Ia8966975c15b9a930f54b4db0fc75f7002dcffe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304461
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40013}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
This CL prepares for send packet batching support in later CLs.
Bug: chromium:1439830
Change-Id: I0bbecfa895aa6d4317ef8049b3789272a440d032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304282
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40009}
rtcp::ReportBlock class is designed for serialization while
ReportBlockData designed for passing report block information across
multiple components.
This slightly reduce how RtcpTransceiver and RtcpReceiver interact with other webrtc components
Bug: webrtc:8239
Change-Id: I582e3d7b32dc6357954b29a1def37e2e72116a74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304285
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40006}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: I580e8412d379931bfdf9517e0a8be25c19e0cd32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304100
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40004}
Per the docs, the caller is responsible for freeing the memory.
Bug: chromium:1441804
Change-Id: I9aaae493a1a86d8ab4f03930715a643a3c9fb61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304061
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39983}
ReportBlockData class is better documented and has wider usage.
Bug: webrtc:13757
Change-Id: Ie5f2275f2f0236267172e6dd1ce5c2dfb2193ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304101
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39980}
