6518 Commits

Author SHA1 Message Date
Danil Chapovalov
c146b5f77b Represent unset VideoPlayoutDelay with nullopt rather than special value
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.

Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta

Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
2023-08-18 13:17:50 +00:00
Danil Chapovalov
31251fa817 Delete deprecated RTPSender::SetCsrcs
Bug: None
Change-Id: I0ce16dc51fa8aa9dcb1f3e96e62f05be11e3cba2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315941
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40566}
2023-08-18 11:50:02 +00:00
Johannes Kron
e5af52e1f2 Make DesktopFrameCGImage::CreateFromCGImage() public
CreateFromCGImage() is needed to be called directly when we move away
from using the deprecated API that is used in CreateForWindow().

Bug: chromium:1471931
Change-Id: I28a2972a2a995103829fd9aff4bc1137a8b424b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315324
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40563}
2023-08-18 01:12:17 +00:00
Philipp Hancke
1e42d83db4 Make VCMReceiveCallback::FrameToRender pure virtual again
after the downstream tests have been updated.

BUG=webrtc:14728

Change-Id: I9cf7eb607f8b26acf985d90625e55bac257a2606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316220
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40557}
2023-08-16 10:44:28 +00:00
Tony Herre
392e4714e7 Remove deprecated TransformableAudioFrameInterface::getHeader() method
Fixed: chromium:1456628
Change-Id: I12ea08070578de846f042c64f2808b55de1603a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315960
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40555}
2023-08-15 16:31:02 +00:00
Harald Alvestrand
d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces.
This CL takes away all usages except for Android code.

Low-Coverage-Reason: Refactoring old code
Bug: webrtc:15410
Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40554}
2023-08-15 13:20:21 +00:00
Philipp Hancke
5165743926 Reland "Fix definition of keyframes decoded statistics"
This is a reland of commit 0e37f5ebd44183d9fe5318d844235aae28fda86a
with backward compability added to allow downstream tests to migrate to the new signature.

Original change's description:
> Fix definition of keyframes decoded statistics
>
> which are defined to be measured after decoding, not before:
>   https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-keyframesdecoded
>
> BUG=webrtc:14728
>
> Change-Id: I0a83dde278e1ebe8acf787bdac729af369a1ecf8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315520
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40545}

BUG=webrtc:14728

Change-Id: I4cf52bb22ba8244155b4fa8c367b9c0306a77590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316120
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40553}
2023-08-15 12:09:46 +00:00
Mirko Bonadei
2dbf6e09c1 Revert "Fix definition of keyframes decoded statistics"
This reverts commit 0e37f5ebd44183d9fe5318d844235aae28fda86a.

Reason for revert: Breaks downstream tests (non backwards compatible change)

Original change's description:
> Fix definition of keyframes decoded statistics
>
> which are defined to be measured after decoding, not before:
>   https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-keyframesdecoded
>
> BUG=webrtc:14728
>
> Change-Id: I0a83dde278e1ebe8acf787bdac729af369a1ecf8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315520
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40545}

BUG=webrtc:14728
No-Presubmit: true
No-Tree-Checks: true
No-Try: true

Change-Id: Idd31fbe6b7173e4bcdfaabfc1704ec6513e80ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315961
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#40550}
2023-08-14 17:12:56 +00:00
Markus Handell
ed2ce81f27 RtpSenderEgress: make potential crash more explicit.
Fixed: chromium:1472676
Change-Id: I1701b65e05c13c95392da90e6642e4eb6fe133ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40548}
2023-08-14 14:54:53 +00:00
Danil Chapovalov
4c17e2dbce Refactor csrcs managment in RtpSender
contributing sources are usually decided per packet, and thus having persistent member for csrcs makes them less natural to use.

Bug: None
Change-Id: I804d58ace574368f8cdd4356a15471110e530744
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291334
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40547}
2023-08-14 13:30:33 +00:00
Andrey Volykhin
6b51e728e6 fec: Skip traversal the list of recovered packets if possible
Do not traverse the list of recovered media packets
if none of them was recovered through FEC recovery procedure.

Bug: None
Change-Id: Ib3aa59c946919fab08f0e20fcf279b1b8032d0e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315320
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Andrei Volykhin <andrey.volykhin@lge.com>
Cr-Commit-Position: refs/heads/main@{#40546}
2023-08-14 13:21:14 +00:00
Philipp Hancke
0e37f5ebd4 Fix definition of keyframes decoded statistics
which are defined to be measured after decoding, not before:
  https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-keyframesdecoded

BUG=webrtc:14728

Change-Id: I0a83dde278e1ebe8acf787bdac729af369a1ecf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315520
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40545}
2023-08-14 12:13:38 +00:00
Danil Chapovalov
24e704f148 Cleanup calculating time between RTCP reports
Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
Use more strict types for the calculation to make it clearer.
Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.

Reland of https://webrtc-review.googlesource.com/c/src/+/315143

Bug: None
Change-Id: I41ce383a2f19d489e4cae0b1bf1f720e0ffdd17a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315460
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40538}
2023-08-10 20:40:15 +00:00
Danil Chapovalov
3b75c39d59 Revert "Cleanup calculating time between RTCP reports"
This reverts commit 762f193ca4cf7f32c665461e92b36811b1ef6fbd.

Reason for revert: breaks downstream test

Original change's description:
> Cleanup calculating time between RTCP reports
>
> Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
> Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
> Use more strict types for the calculation to make it clearer.
> Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.
>
> Bug: None
> Change-Id: Ie8c6b9720095cd1cc3f9814b9df16700119337c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315143
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40529}

Bug: None
Change-Id: I8c83013523120a84f236e8efa0d122363e7a228b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315381
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40535}
2023-08-10 14:18:52 +00:00
Danil Chapovalov
490f0b82d7 Cleanup usage of csrcs in RtpSenderVideoFrameTransform
CSRCs are decided on a per frame bases, thus keeping a constant copy of
csrcs inside the rtp sender transform delegate is confusing: when transform delegate is created, csrcs list is always empty.

Bug: None
Change-Id: Id94acc76857a47ad9a1dd8254648ab9cb5d6d31d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311840
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40533}
2023-08-10 10:30:29 +00:00
Palak Agarwal
86162d94d5 Implement setMetadata for receiver encoded video frames
This change adds a new function to RTPFrameObject to allow setting the
RTPVideoHeader from VideoFrameMetadata.

The setMetadata function in TransformableVideoReceiverFrame disallows
changing anything other than frameID and dependencies.

Change-Id: I74e55ffbe1f426b660c2e243b20358c6a6cc2ffd
Bug: chromium:1464853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314963
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40530}
2023-08-09 10:01:58 +00:00
Danil Chapovalov
762f193ca4 Cleanup calculating time between RTCP reports
Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
Use more strict types for the calculation to make it clearer.
Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.

Bug: None
Change-Id: Ie8c6b9720095cd1cc3f9814b9df16700119337c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315143
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40529}
2023-08-09 09:14:41 +00:00
Philipp Hancke
240e783d7f Stop using invalid payload type 200 in audio/red unit test
and fix the follow-up mistake in the test

BUG=None

Change-Id: Id7a20769cc1d03dd8154564f948e8138ff8c4e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315220
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40528}
2023-08-09 08:13:52 +00:00
Danil Chapovalov
277fb3cd0e Cleanup RtpSender unittest to use Timestamp instead of plain int
Bug: webrtc:13757
Change-Id: I0893805a8e4bc66c0c6cff232576387c2ee60b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315182
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40525}
2023-08-08 14:07:45 +00:00
Philipp Hancke
82e5f91a2b audio: fix handling of RED packets where the primary encoding is too large
by falling back to the primary encoding. This can happen with
opus stereo packets at the maximum bitrate which results in
1276 encoded bytes.

BUG=chromium:1470261

Change-Id: I3fd9bb30773963a519bbb5da44fe71db5dec2bd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315141
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40524}
2023-08-08 13:40:26 +00:00
Harald Alvestrand
34d82df2ba Use ArrayView versions of SendRtp and SendRtcp
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.

Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
2023-08-07 08:28:48 +00:00
Danil Chapovalov
ebb2bfd239 Delete deprecated member CodecSpecificInfoVP9::end_of_picture
This field is unused. It was deprecated in
https://webrtc-review.googlesource.com/c/src/+/192542

Bug: None
Change-Id: Iba264c03e16fc240893595075b525d44efa8ab96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314720
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40504}
2023-08-02 14:05:31 +00:00
Sergey Silkin
86a7969a6d Synchronize access to callbacks map
Bug: webrtc:14852
Change-Id: I65a608976056865849f4175411febc8093402de1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40500}
2023-08-02 11:38:25 +00:00
Danil Chapovalov
9a09ed73c5 Cleanup RemoteBitrateEstimatorSingleStream
Store Detectos in a map by value instead of by unccessary pointer

Bug: None
Change-Id: Iab9904aafca02d9f9ae6633c87de860a5bd62ac7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313621
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40499}
2023-08-02 10:11:38 +00:00
Danil Chapovalov
4f4e989436 In remote bitrate estimator pass packet using RtpPacketReceived class
Bug: webrtc:15054
Change-Id: I23c9008e1979a56bba9421a00e4f0f8ff937d122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40485}
2023-07-28 10:22:16 +00:00
Danil Chapovalov
920abcc9bc In RtpSenderVideo::UpdateConditionalRetransmit use typed time and framerate instead of plain ints
Bug: webrtc:13757
Change-Id: If2df5418dacd2b95387fa74a9bc226426b207aee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313041
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40483}
2023-07-27 14:35:42 +00:00
Danil Chapovalov
ac412a4ee3 In RTPSenderVideo delete deprecated variants of functions to send video frame
Bug: webrtc:13757
Change-Id: I0bef9cc17e599382cc2265d61a2a538f2534356f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312860
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40472}
2023-07-25 10:47:47 +00:00
Danil Chapovalov
7b42f35bcc Remove artifical extra RTP packet capacity
Instead allow RtpPacket to exceed configured capacity when setting payload

Bug: None
Change-Id: I02fc080ffa3127ffbe0dade1f200dd7456a6a128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40471}
2023-07-25 06:33:09 +00:00
Harald Alvestrand
7dbf55437f Ensure payload type frequency does not cause divide-by-zero
This CL does 2 things:
- Change the DCHECK for payload_type_frequency to a CHECK (so that
this error will be a crash not a divide-by-zero)
- Change the replay helper that was used by the fuzzer to set the
frequency of the packets to the video value (90K).

Bug: chromium:1466826
Change-Id: I39941f250b1782b36a3bcddfd347a016591466ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312700
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40468}
2023-07-24 16:06:08 +00:00
Danil Chapovalov
950e231b63 In RtpRtcp use BitrateTracker instead of RateStatistics to measure bitrate
BitrateTracker uses RateStatistics underneath, thus algorithm is the same,
but it provides Timestamp/TimeDelta friendly interface

Bug: webrtc:13757
Change-Id: I9f2fcb3d498b2a137b531b94b660d15aa273c4bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40465}
2023-07-24 14:57:29 +00:00
Jan Grulich
666d707450 Video capture PipeWire: guard callback to avoid concurrent access
Make sure the callback is reset when tearing down the PipeWireSession
and that there is no concurrent access to it, which can potentially lead
to a crash.

Bug: webrtc:15386
Change-Id: I0b09002fe0479dc1cd946c80684bcc5d8754d54a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311546
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40464}
2023-07-24 14:48:33 +00:00
Harald Alvestrand
00f11224fd Remove extra usage of video-content-type header extension
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.

Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
2023-07-22 21:47:08 +00:00
Caroline Liu
0689cfc6ce Reland "[fuchsia] remove Scenic/UseFlatland dependency in DesktopCapturer"
This reverts commit 726992d7a4722b8a283d797d04432d0c6335ca96.

Reason for revert: Relanding with original errors fixed (tested by building the patch locally against Chromium)

This change no longer attempts to migrate the display size protocol from fuchsia.ui.scenic.Scenic/GetDisplayInfo to fuchsia.ui.display.singleton.Info/GetMetrics because the latter API was introduced in Fuchsia API 12, which is not yet supported in Chrome (hence some of the build errors causing the revert).

Original change's description:
> Revert "[fuchsia] remove Scenic and GFX  dependencies in DesktopCapturer"
>
> This reverts commit fe5be2eb4ff8dccd96257fb8cbf32500c636c358.
>
> Reason for revert: This breaks the WebRTC roll into Chromium:
>
> - https://chromium-review.googlesource.com/c/chromium/src/+/4688561
> - https://ci.chromium.org/ui/p/chromium/builders/try/fuchsia-binary-size/399140/overview
>
> Error:
>
> [4273/4389] CXX obj/third_party/webrtc/modules/desktop_capture/desktop_capture/screen_capturer_fuchsia.o
> FAILED: obj/third_party/webrtc/modules/desktop_capture/desktop_capture/screen_capturer_fuchsia.o
> ../../buildtools/reclient/rewrapper -cfg=../../buildtools/reclient_cfgs/chromium-browser-clang/rewra...(too long)
> ../../third_party/webrtc/modules/desktop_capture/screen_capturer_fuchsia.cc:59:10: error: use of undeclared identifier 'capturer'
> 59 |   return capturer(new ScreenCapturerFuchsia());
> |          ^
> ../../third_party/webrtc/modules/desktop_capture/screen_capturer_fuchsia.cc:199:36: error: no type named 'InfoSyncPtr' in namespace 'fuchsia::ui::display::singleton'
>
> Original change's description:
> > [fuchsia] remove Scenic and GFX  dependencies in DesktopCapturer
> >
> > We previously used:
> > - fuchsia.ui.scenic.Scenic/UsesFlatland to determine whether to use
> >   Flatland; from now on it should always be the case, so this check is
> >   no longer necessary.
> > - fuchsia.ui.scenic.Scenic/GetDisplayInfo to get
> >   fuchsia.ui.gfx.DisplayInfo. This has been migrated to
> >   fuchsia.ui.display.singleton.Info/GetMetrics and
> >   fuchsia.ui.display.singleton.Metrics.
> >
> > Bug: fuchsia:100303
> > Change-Id: I147da9ffdf0ca49e1c5bde5d188e434fc660becc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311860
> > Reviewed-by: Emircan Uysaler <emircan@google.com>
> > Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> > Commit-Queue: Caroline Liu <carolineliu@google.com>
> > Cr-Commit-Position: refs/heads/main@{#40432}
>
> Bug: fuchsia:100303, b/291393959
> Change-Id: Iae70e568a8c9819e40e48069af8cea0d4ef2b6c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311801
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40436}

Bug: fuchsia:100303, b/291393959
Change-Id: Icb7074ac86c1804ab2bdf809ea1496539ee2bf80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312000
Commit-Queue: Caroline Liu <carolineliu@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40452}
2023-07-21 16:47:17 +00:00
Danil Chapovalov
630c40d716 Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types
Bug: webrtc:13757
Change-Id: I2f21b14ecf003c5cb0c4c92d0c6b9b6f11c35f71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311945
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40450}
2023-07-21 10:36:49 +00:00
henrika
e66a85c278 kDummyAudio now also creates Dummy ADM on Android
The old Android ADM was removed in https://webrtc-review.googlesource.com/c/src/+/271841.

This change resulted in a NULL as result when asking for a
kDummyAudio ADM on Android.

The small change below should ensure that a dummy ADM can be
created on Android as well.

Bug: webrtc:7452, b/291275589
Change-Id: I2c995ce6ba9a4117e3e39596546b133fe1c49204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311946
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40440}
2023-07-17 15:22:22 +00:00
Björn Terelius
fd5bdca28c Revert "Disable VideoCaptureTest due to flakyness"
This reverts commit 4ee5e5f294dcd342851bbe80004ddd661fc6f3ed.

Reason for revert: HW fixed (hopefully)

Original change's description:
> Disable VideoCaptureTest due to flakyness
>
> Bug: webrtc:15229
> Change-Id: I3303b13be74d9eae5c52ecb2b920c23ac7d063d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308220
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40244}

Bug: webrtc:15229
Change-Id: I30ad37236ffcb56d7ffe4d3efa3d03705be25c47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311804
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40437}
2023-07-17 09:02:44 +00:00
Mirko Bonadei
726992d7a4 Revert "[fuchsia] remove Scenic and GFX dependencies in DesktopCapturer"
This reverts commit fe5be2eb4ff8dccd96257fb8cbf32500c636c358.

Reason for revert: This breaks the WebRTC roll into Chromium:

- https://chromium-review.googlesource.com/c/chromium/src/+/4688561
- https://ci.chromium.org/ui/p/chromium/builders/try/fuchsia-binary-size/399140/overview

Error:

[4273/4389] CXX obj/third_party/webrtc/modules/desktop_capture/desktop_capture/screen_capturer_fuchsia.o
FAILED: obj/third_party/webrtc/modules/desktop_capture/desktop_capture/screen_capturer_fuchsia.o
../../buildtools/reclient/rewrapper -cfg=../../buildtools/reclient_cfgs/chromium-browser-clang/rewra...(too long)
../../third_party/webrtc/modules/desktop_capture/screen_capturer_fuchsia.cc:59:10: error: use of undeclared identifier 'capturer'
59 |   return capturer(new ScreenCapturerFuchsia());
|          ^
../../third_party/webrtc/modules/desktop_capture/screen_capturer_fuchsia.cc:199:36: error: no type named 'InfoSyncPtr' in namespace 'fuchsia::ui::display::singleton'

Original change's description:
> [fuchsia] remove Scenic and GFX  dependencies in DesktopCapturer
>
> We previously used:
> - fuchsia.ui.scenic.Scenic/UsesFlatland to determine whether to use
>   Flatland; from now on it should always be the case, so this check is
>   no longer necessary.
> - fuchsia.ui.scenic.Scenic/GetDisplayInfo to get
>   fuchsia.ui.gfx.DisplayInfo. This has been migrated to
>   fuchsia.ui.display.singleton.Info/GetMetrics and
>   fuchsia.ui.display.singleton.Metrics.
>
> Bug: fuchsia:100303
> Change-Id: I147da9ffdf0ca49e1c5bde5d188e434fc660becc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311860
> Reviewed-by: Emircan Uysaler <emircan@google.com>
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Caroline Liu <carolineliu@google.com>
> Cr-Commit-Position: refs/heads/main@{#40432}

Bug: fuchsia:100303, b/291393959
Change-Id: Iae70e568a8c9819e40e48069af8cea0d4ef2b6c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311801
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40436}
2023-07-17 08:12:48 +00:00
Caroline Liu
fe5be2eb4f [fuchsia] remove Scenic and GFX dependencies in DesktopCapturer
We previously used:
- fuchsia.ui.scenic.Scenic/UsesFlatland to determine whether to use
  Flatland; from now on it should always be the case, so this check is
  no longer necessary.
- fuchsia.ui.scenic.Scenic/GetDisplayInfo to get
  fuchsia.ui.gfx.DisplayInfo. This has been migrated to
  fuchsia.ui.display.singleton.Info/GetMetrics and
  fuchsia.ui.display.singleton.Metrics.

Bug: fuchsia:100303
Change-Id: I147da9ffdf0ca49e1c5bde5d188e434fc660becc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311860
Reviewed-by: Emircan Uysaler <emircan@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Caroline Liu <carolineliu@google.com>
Cr-Commit-Position: refs/heads/main@{#40432}
2023-07-14 21:05:29 +00:00
Philipp Hancke
c0ed83eac2 Bail out early if the RTP send module for a SSRC was not found
since it might have been deregistered previously.

BUG=chromium:1454860,chromium:1459124

Change-Id: I70ba43265361d040e568f83b6400ff8f3c2a8e98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40431}
2023-07-14 10:45:36 +00:00
Danil Chapovalov
aa8faa6423 Update RtcpReceiver to use Timesetamp/TimeDelta types instead of raw ints
Bug: webrtc:13757
Change-Id: Ie0317a584406bec3c34403a7bc8059e4272b339f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311674
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40429}
2023-07-13 12:08:10 +00:00
Jan Grulich
0e9556a90c Desktop capture: introduce capturer requesting both screen and windows
When PipeWire and xdg-desktop-portals are used, we can actually combine
both source types into one request. Make this part of the API for those
who want to use it this way, e.g. Firefox or Electron, otherwise they
will end up making two simultaneous requests, resulting into two dialogs
at the same time asking, while they can be combined into just one.

Bug: webrtc:15363
Change-Id: Ib6e1e47f66cb01d5c65096aec378b44c3af5f387
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311549
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40425}
2023-07-12 18:59:24 +00:00
Anne Redulla
73d51f8e84 [ssci] Added Shipped field to READMEs
This CL adds the Shipped field (and may update the
License File field) in Chromium READMEs. Changes were
automatically created, so if you disagree with any of
them (e.g. a package is used only for testing purposes
and is not shipped), comment the suggested change and
why.

See the LSC doc at go/lsc-chrome-metadata.

Bug: b:285450740
Change-Id: If4955c6f6e7b58e0c99469fc45ed5b9e8f30a32b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311720
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Anne Redulla <aredulla@google.com>
Cr-Commit-Position: refs/heads/main@{#40424}
2023-07-12 07:31:06 +00:00
Jianhui Dai
32a8169a65 Use common VideoFrameTypeToString helper
This CL cleans up all local conversions, in favor of the common helper
function.

Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
2023-07-12 00:28:47 +00:00
Alfred E. Heggestad
6a4f409241 pacing_controller: add millisecond units to warning message
Bug: None
Change-Id: I8a5c5ca6a641a74213116a734f3c19c6972e5916
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311400
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40418}
2023-07-11 17:00:06 +00:00
Tony Herre
9d677f4cdc Set surrogate receive times for transformed sender frames
Without this, 'Sender' frames inserted into the writer of an encoded
transform have an invalid receive time (0), which breaks all later
heuristics which build on the receive time, eg the VCMTiming estimators
used for controlling the playback delay.

Bug: chromium:1463451
Change-Id: I413c884e08986148d4a854cd275212b21d093ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311544
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40416}
2023-07-11 14:30:18 +00:00
Jan Grulich
8fcc6df79d PipeWire capturer: increase buffer size to avoid buffer overflow
Recently added framerate option can cause a buffer overflow and make
PipeWire to fail on negotiation, which effectively makes screen sharing
not to work.

Bug: webrtc:15346
Change-Id: I4a68e26c8f85ca287b06a25da500b6a7009e075f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311541
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40413}
2023-07-10 17:32:30 +00:00
Artem Titov
599367595d Allow StartRecording if capturer is null in test ADM
Bug: b/272350185
Change-Id: I3aca6d8b3eb4fd39a6d39f1fea272858e18193bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311463
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40407}
2023-07-07 14:01:38 +00:00
Alfred E. Heggestad
e9ae738c7c rtcp_sender: uppercase protocol names (RTCP) in log messages
Bug: None
Change-Id: Ie6683897fca469a15c1aa054eeb1b2d378b22bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311461
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40405}
2023-07-06 13:34:31 +00:00
Philipp Hancke
3f10b4917e Log SSRC for video decode errors
which makes it possible to grep the logs for all decode errors
on a particular SSRC.

BUG=None

Change-Id: I4aa54434f0b85932313adaf39e099729991a4700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308823
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40404}
2023-07-06 11:24:16 +00:00
Artem Titov
1a8c1aedbc Add raw file audio capturer/renderer for test ADM
Bug: b/272350185
Change-Id: Ie8c7f7be30d06b238240086eee172332287c77ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311280
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40399}
2023-07-04 11:03:25 +00:00