This is a reland of 3097008de03b6260da5cfabb5cbac6f6a64ca810
Patchset 1 is a pure reland. Patchset 2 contains a bugfix plus a test
covering that case.
Bug: webrtc:12354, chromium:1230448
Original change's description:
> Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields.
>
> These fields will be used for bitstream validation in upcoming CLs.
> A new vp9_constants.h file is also added, containing common constants
> defined by the bitstream spec.
>
> Bug: webrtc:12354
> Change-Id: If04256d83409069c8bee43ad41aed41c3707dfd3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226060
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34476}
Bug: webrtc:12354
Change-Id: Ibd301eb458a6104b562cefbc0e616c39b54fb38b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229060
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34789}
We already default to PipeWire 0.3 and there is no reason to keep
continue supporting an old version of PipeWire which is not maintained
anymore, wont't get any update or new features. It also makes the code
easier to understand since we can remove all ifdefs we had to support
two versions simultaneously.
Bug: chromium:1146942
Change-Id: I7156e1784ebfad111485a2944199563568a75eec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227345
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34765}
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings
Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
The config flag will be removed once downstream usage is gone.
Bug: webrtc:11340
Change-Id: Iee8816660009211540d9b09bb3cba514455d709b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228431
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34757}
Since https://webrtc-review.googlesource.com/c/src/+/228433 both audio
and video now only call Get/SetRtpState while not registered to the
packet router.
We can thus remove the lock around packet sequencer and just use a
thread checker.
Bug: webrtc:11340
Change-Id: Ie6865cc96c36208700c31a75747ff4dd992ce68d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228435
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34755}
As a first step we only want to enable frame pacing for the case
where min playout delay == 0 and max playout delay > 0.
Bug: chromium:1237402, chromium:1239469
Change-Id: Icf9641db7566083d0279135efa8618e435d881eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228640
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34752}
With deferred packet sequencing, the PacketSequencer instance is called
directly from the RtpRtcp module while before it was called from within
the RTPSender while holding a lock.
Since sequence number assignment happens on the same thread as actual
packet sending, though thought was that locking was no longer needed.
Unfortunately, SetRtpState()/GetRtpState() also exists - and while they
should only be called on creating/destruction there is a possible race
where a delayed packet from the pacer accesses the sequencer while
GetRtpState() is being called.
For now, this CL just adds a lock to guard sequencer. Follow-ups will
make sure get/set state is never called while module is attached to
the packet router. After that, the lock can be removed again.
Bug: webrtc:11340, webrtc:12470
Change-Id: I123c762fb4afd20b3a6bd03b86234eb9ec34a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228430
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34723}
This is a reland of 320c57b7c6bf68d8612a4f135f44b3d29e802113
Original change's description:
> red: remove special-casing of no-redundancy
>
> removes the special-casing of not sending a RED header when there is no redundant payload.
> This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
>
> BUG=webrtc:11640
>
> Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34703}
Bug: webrtc:11640
Change-Id: I5e5687be575183ee16d74df4a8170e4fedad739f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228422
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34713}
This reverts commit 320c57b7c6bf68d8612a4f135f44b3d29e802113.
Reason for revert:
Breaks CI tests: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux32%20Release/27236/overview
All CI tests: https://ci.chromium.org/p/webrtc/g/ci/console
Error is of the following type:
```
../../modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc:195: Failure
Expected equality of these values:
1u
Which is: 1
encoded_info_.encoded_bytes
Which is: 2
Stack trace:
0x56b298d9: webrtc::AudioEncoderCopyRedTest_CheckPayloadSizesSingle_Test::TestBody()
0x572fe317: testing::internal::HandleExceptionsInMethodIfSupported<>()
0x572fe1d4: testing::Test::Run()
0x572ff2ee: testing::TestInfo::Run()
```
Original change's description:
> red: remove special-casing of no-redundancy
>
> removes the special-casing of not sending a RED header when there is no redundant payload.
> This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
>
> BUG=webrtc:11640
>
> Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34703}
TBR=henrik.lundin@webrtc.org,devicentepena@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Ide409232720df32b24022f99228f3b6ae81f06fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228421
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34707}
When pacing is enabled for the low latency rendering path,
frames are sent to the decoder in regular intervals. In case of a
jitter, these frames intervals could add up to create a large latency.
Hence, disable frame pacing if the pre-decode queue grows beyond the
threshold. The threshold for when to disable frame pacing is set
through a field trial. The default value is high enough so that
the behavior is not changed unless the field trial is specified.
Bug: chromium:1237402
Change-Id: I901fd579f68da286eca3d654118f60d3c55e21ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228241
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34705}
removes the special-casing of not sending a RED header when there is no redundant payload.
This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
BUG=webrtc:11640
Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34703}
fixes an incorrect redundancy shift and add tests that would have caught this bug.
BUG=webrtc:11640
Change-Id: I6fe2fb21587fffc5fee4d403ac898e12d525a1cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224120
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34702}
Denormal numbers (see [1]) may origin in APM when the input is zeroed
after a non-zero signal. In extreme cases, instructions involving
denormal operands may run as much as 100 times slower, which seems to
be the case (to some extent) of crbug.com/1227566.
This CL adds a class that disables denormals only via hardware on x86
and on ARM. The class is used in APM and it is an adaption of [2].
Tested: appr.tc call on Chromium (Win, Mac)
[1] https://en.wikipedia.org/wiki/Denormal_number
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/platform/audio/denormal_disabler.h
Fixed: chromium:1227566
Change-Id: I0ed2eab55dc597529f09f93c26c7a01de051fdbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227768
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34701}
Caches the TCP fairness limit to avoid redundant calculation. Adds option to append the delay based estimate as a candidate. Makes the appending of acknowledged bitrate as a candidate optional. Adds a log-bandwidth bias term.
(submit on behalf of crodbro)
Bug: webrtc:12707
Change-Id: Ic4b0f58e6f0bc3b117fe78a2321a07db65afd9dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228163
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34687}
Remove private members that are no longer used or always have same value
Use less allocations
Bug: None
Change-Id: I5430c2356f0039212baf8b248b92775e8852ce1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227765
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34665}
With this turned on, packets will be sequence number after the pacing
stage rather that during packetization.
This avoids a race where packets may be sent out of order, and paves
the way for the ability to cull packets from the pacer queue without
causing sequence number gaps.
For now, the feature is off by default. Follow-ups will enable it for
video and audio separately.
Bug: webrtc:11340, webrtc:12470
Change-Id: I6d411d8c85b9047e3e9b05ff4c2c3ed97c579aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208584
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34661}
Schedule the frames to be decoded based on the pacing delay from the
last decode scheduled time. In the current implementation, multiple
threads and different functions in same thread can call
MaxWaitingTime(), thereby increasing the wait time each time the
function is called. Instead of returning the wait time for a future
frame based on the number of times the function is called, return the
wait time only for the next frame to be decoded. Threads can call the
function repeatedly to check the waiting time for next frame and wake
up and then go back to waiting if an encoded frame is not available.
Change-Id: I00886c1619599f94bde5d5eb87405572e435bd73
Bug: chromium:1237402
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226502
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34660}
