This is a reland of commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I123d1134a212f65cfbc90ecec9013d0aafebd9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308721
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40294}
This is useful in environments where OpenSSL may not be available.
Bug: webrtc:15240
Change-Id: I7ba29e44bd1d25231df13ee79dacc74f260ded67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40293}
while not really covered by
https://www.rfc-editor.org/rfc/rfc5576.html#section-4.2
and using the same SSRC for RTX and primary payload may work
since payload type demuxing *could* be used is not a good idea.
This also applies to flexfec's FEC-FR.
For the nonstandard SIM ssrc-group duplicates make no sense.
This rejects duplicates for unknown ssrc-groups as well.
BUG=chromium:1454860
Change-Id: I3e86101dbd5d6c4099f2fdb7b4a52d5cd0809c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308820
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40292}
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}
Also add a preprocessor definition to avoid redefinition in downstream projects.
Bug: webrtc:15241
Change-Id: Ic55d98c3d3a69b9b19195ee78f03af6e38fdd0e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40289}
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
with a single encoding or multiple encodings. As long as only one
encoding is active we get a single L1Tx ssrc, same as legacy API.
Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.
The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.
This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.
Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
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}
This reverts commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc.
Reason for revert: Breaks downstream projects
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I57778cccc3a13eb9f955f6ece054dee0ff5a7e92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308720
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40278}
In M113 we made it possible to opt-in to spec-compliant VP9 using
scalabilityMode and scaleResolutionDownBy. Since this would change
behavior in some edge cases a kill-switch flag was also added.
It turns out it was not needed (current Stable: M114) so we can remove
the flag.
Bug: webrtc:14884
Change-Id: Ie3006164c4d6e90acad1d1f4df2fe2b6e3cb2c35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308683
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40277}
This allows simplifying code in the codebase to be able to remove a lot
of templated code and special casing for either AudioCodec and VideoCodec.
Code simplifications will come in later changes.
Bug: webrtc:15214
Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40276}
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}
This CL adds a new video encode tool that supports to encode video at
specified codec, scalability mode, resolution, frame rate, bitrate,
key frame interval and the number of encoding frames.
The video encoder accepts video frames from `FrameGeneratorInterface`,
and supports `SquareFrameGenerator`, `SlideFrameGenerator` and
`IvfFileFrameGenerator`.
All the encoded bitstreams are wrote into ivf output files.
The purposes of this video encoder tool are:
1. Check the functionalities of video codecs and scalability modes.
2. Optimize video quality at different encode setting.
3. Fine tune the bitrate controller.
4. Compare the quality of different codecs at the same setting.
5. And more.
TESTS: Run the tool at 1280x720, 30fps, 2000kbps, 100 GOP, 300 frames:
vp8 [L1T1 L1T3]
h264 [L1T1 L1T3]
vp9 [L1T1 L1T3 L3T3_KEY]
av1 [L1T1 L1T3 L3T3_KEY]
Bug: webrtc:15210
Change-Id: I3b0e463cf3236cd9a481fbab5688643c203958da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307361
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40267}
into separate sections for each implemented class.
Bug: webrtc:13931
Change-Id: I600f49f3fb195761d13d304f112f36c7c62689df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308120
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40260}
This can be done now as the function SetRTPTimestamp is now overriden
in blink MockTransformableAudioFrame.
Change-Id: I4fa4cb81d0282fea864818f0f2d9a5ed881a5d30
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40257}
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}
Pushing it to the list of extensions to negotiate could result
in enabling it in production.
BUG=None
Change-Id: I98599e9fbac7e2b81b3f2ad0c7759bb052d9d9d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306101
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40250}
To avoid downstream repositories having to deal with a roller breakage
on every major revision to the Android NDK, use an unversioned CIPD
path. Versioned CIPD paths cause downstream roller scripts to assume
each version is an unrelated package and requires manual intervention.
Bug: chromium:1446443
Change-Id: I40e8ecec5a451a1ec754c04b35fa2c26519dd528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308320
Auto-Submit: Prashanth Swaminathan <prashanthsw@google.com>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Prashanth Swaminathan <prashanthsw@google.com>
Cr-Commit-Position: refs/heads/main@{#40249}
This CL implements {,Logging}DelayVariationCalculator, whose purpose is to calculate simple inter-arrival metrics for a sequence of RTP frames. Uses could include RtcEventLog analysis and ad hoc testing.
Want lgtm: asapersson
Bug: webrtc:15213
Change-Id: I3f9d13a2c4fa66b6f1229c1b6fcd66a6911070de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306741
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40247}
Now that chromium/base has rolled and switched the android_ndk_root to
the new android_toolchain directory, remove the stale Android NDK. Also
update the license generation and build helper scripts to remove
references to the previous NDK.
Bug: chromium:1448383
Test: Verified build of WebRTC.
Change-Id: Ic2b6009f454d67da60231bbcbb5c27bde8407ef3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307541
Commit-Queue: Prashanth Swaminathan <prashanthsw@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40246}
