If we don't pass the frames to OnFrameEncoded, we can't see the frames
being sent in the SVC tests. We want to check the frames even if the
SFU would discard them later.
Bug: webrtc:11607
Change-Id: I5b9c6a86c0966047efa7be088f90e83e01f7900b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273350
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38159}
New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
To reduce usage of rtc::MessageHandler, hide rtc::Thread::Send into private section with intention to deprecate it in favor of the new Invoke function.
Bug: webrtc:9702, webrtc:11318
Change-Id: Ib4c26f9abc361e05a45b2a91929af58ab160b3f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274166
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38036}
This is unused at the moment and webrtc::SimulatedNetwork is going
through a refactoring, to keep things simple and well tested this CL
removes CoDel but nothing blocks us from re-implementing it when needed.
No-Try: True
Bug: webrtc:14426
Change-Id: Ie7d40d20a66d3939fc7d3251c47e4f13f3869a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274407
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38032}
The value of VP9().interLayerPred isn't used in the VP9 encoder
when scalability_mode is present.
Bug: webrtc:11607
Change-Id: I2ce606e5a91dfe087f652763cbcc258db0156f5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273400
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37937}
Fix reporting of time_between_freezes_ms when all peers were
unregistered before end of the call.
Bug: b/243501613
Change-Id: Ie2b77d8399e4f4bf672e80f62ed27609336af171
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272650
Reviewed-by: Andrey Logvin <landrey@google.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37885}
This CL adds TimeDelta support to the rtc::Event, and updates
the Wait implementations to work with the improved precision.
Bug: webrtc:14366
Change-Id: Iefeb638b18176a34f4ed2a5131754a7b7e6c9e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272002
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37831}
This was only set to nullptr in non-test environments and was thusly
unused. With this change, the stats callbacks are gaurenteed to only
come from the VideoStreamBufferController and so the thread checks can
be removed.
Bug: webrtc:14003
Change-Id: Iaf0e77aa7c45a317e38ae27739edcefd3123d832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37816}
Moves FrameBuffer2 to its own GN target to reduce the binary size of the
video target.
Bug: None
Change-Id: I40e86a1eabc0c9e8e6fada3dcdb4e3a043c61c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271286
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37803}
It no longer has to interact with the decode queue, that will only
happen in VideoReceieveStream2. This moves some members in
VideoReceieveStream2 to the packet sequence which removes a few
post-tasks.
Bug: webrtc:14003, webrtc:11993
Change-Id: I4641b593b1a2f68e017c384b73ee4e75d06cf559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271700
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37802}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Using ms() was converting the value to an int before putting it into a
double, causing the microseconds to be dropped. This has the most impact
on decode time metrics which are ofter less than 1ms.
Bug: webrtc:14339
Change-Id: Ie8401ba5a46eb3b35e8a699acfdad2dcd32a8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271163
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37734}
The data that's used to report the histograms is owned by UlpfecReceiver
and moving the reporting there, simplifies things as configuration
changes happen in RtpVideoStreamReceiver2 (which currently require all
receive streams to be deleted+reconstructed).
Additional updates:
* Consistently using `Clock` for timestamps. Before there was
a mix of Clock and rtc::TimeMillis.
* Update code to use Timestamp and TimeDelta.
Bug: none
Change-Id: I89ca28ec7067a49d6b357315ae733b04e7c5a2e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271027
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37729}
Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
This CL propose a new API for video dumps in PCLF also removing
differences between p2p and multipeer usage of API.
Bug: b/240540206
Change-Id: Id4d32cc98250500949b3f9e2cf2e86c4bdce7efb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37665}
Replace mock implmentation with manual noop implementaion.
libvpx interface is called a lot, and mock implementation of it adds
noticable overhead.
Bug: chromium:1281020
Change-Id: I7fe5cbfd08d5056a14d75e009acff368700c26a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269214
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37656}
WebRTC doesn't produces such packet and ignores it when receive.
Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
This enables the possibility to use these flags from other main
functions.
Bug: b/217701950, b/238157120
Change-Id: I6e9f09c4be39b2ac690c8e6b6da380ef3c81f460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269140
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37582}
