This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.
Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
wanted_fps_ seems redundant with target_capture_fps_.
The problem with wanted_fps_ is that it lowers the capture fps but does not decimate frames so that a 30 fps stream played at 5 fps is played slowly instead of played at the normal speed with dropped frames.
Change-Id: I1440953f9909ad1d4a102a0671fe933d95498a1f
Bug: b/355120692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42670}
Currently this class assumed that if the same RTP sequence number is unwrapped again result would be the same.
That might not be true when several packets were inserted in between these two calls and unwrapper changed its state
This CL propose instead to unwrap once, and save the result in the intermediate struct.
To minimize the change and the risk, only redundant unwrapping is replaced to use unwrapped sequence number
Bug: webrtc:353565743
Change-Id: I8a18c8c206a0e16010951cabcf81dd9cb1588eda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42662}
In preparation for upcoming changes in GetSimulcastConfig(), which will require a vector of stream resolutions instead of just the max resolution as an input, switch tests to use CreateEncoderStreams() instead of calling GetSimulcastConfig() directly.
Bug: webrtc:351644568, b/352504711
Change-Id: I541dd54a21a8b75028cff07a250f858a47898223
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357400
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42648}
This is a cleanup of simulcast.cc. max_qp is not needed to decide simulcast config. Move setting of max QP in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams(), where it can be set per stream.
Bug: webrtc:351644568, b/352504711
Change-Id: Ia0e3e9d90032383574dc8867b30d362e9c5df7e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42634}
This is a cleanup of simulcast.cc. bitrate_priority is not needed to decide simulcast config. Move setting of bitrate priority in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams().
Bug: webrtc:351644568
Change-Id: I002d728ccf8d141fe4bbb32b390129ce57c830cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357101
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42629}
MakeAudioEncoder planned to be removed, and Create planned to become pure virtual
While at it, cleanup nearby mock usage:
Remove ON_CALL that by default return default constructed result
Remove EXPECT_CALL().Times(AnyNumber()) for a NiceMock
Remove parameters in EXPECT_CALL when all are wildcard
Remove redundant get to deference a smart pointer
Bug: webrtc:343086059
Change-Id: Ica90a4980350cb82bcebd11df6c63a01b828bb9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356884
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42622}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
Some dependencies still exist but are a bit more complex to remove.
This CL removes either unused or easily replaced with ToString()
instances of ostream usage. In one case, moving the operator<<
implementation to the one test file that requires it.
Bug: webrtc:8982
Change-Id: Ia5c840b12a42893494af401317a3daf2fe50ba9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356240
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42582}
The temporal id must be read from `EncodedImage` rather than codec
specifics for AV1. Furthermore, in some configs the spatial id of
`EncodedImage` is populated and set to 0 while the simulcast id can
also be simultaneously populated and set to values, including non-zero.
To solve this, just take the max of the two.
Bug: b/349561566
Change-Id: I46c61b7f0fff7a7ab8d7262c3a8d413f49b3286a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355904
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42573}
This target then will be filled with writer related part from "video_test_support"
This allows downstream to migrate on the new target keeping dependency on the old one.
Bug: None
Change-Id: Ie0b2f0ff9c7896c70b9a204ffedf15afac43c143
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355580
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42539}
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default
Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.
Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
If the task queue is blocked, there is a risk that delay becomes negative. Therefore, use max of calculated time to next schedule and 0.
Bug: webrtc:42224804
Change-Id: Ibae9000192d5042cf62b46d93e8364b58dae0d82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354880
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42501}
Before the schedule starts an absl::AnyInvocvable is executed every time
a packet is enqued. The incocable should return true, if the schedule should
be started.
The pupose is to allow tests to not start a schedule until ICE and DTLs
is connected.
Bug: webrtc:42224804
Change-Id: I61bd63508830f7c27d86f982299ce2be180ff460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354464
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42479}
and move usages to webrtc::RefCountInterface
This CL also moves more stuff to webrtc:: and adds backwards
compatible aliases for them.
Bug: webrtc:42225969
Change-Id: Iefb8542cff793bd8aa46bef8f2f3c66a1e979d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353720
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42446}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
This reverts commit 06815534d2da29a7f41cad2eaab6d2103f0138c2.
Reason for revert: Seems to break importer...
Original change's description:
> Add SchedulableNetworkBehavior and tests.
>
> This is a network behaviour that can change its parameters over time as specified with a schedule proto.
>
> Bug: webrtc:14525
> Change-Id: Idd34cc48c8e3e8311975615f2c3dc3ffb522a708
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352140
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42390}
Bug: webrtc:14525
Change-Id: I4386ffb7629198c74249e416076cab3b4c23a79b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352501
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42391}
This is a network behaviour that can change its parameters over time as specified with a schedule proto.
Bug: webrtc:14525
Change-Id: Idd34cc48c8e3e8311975615f2c3dc3ffb522a708
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352140
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42390}
with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target.
BUG=webrtc:339300437
Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42362}
The test tests that a route change does not cause BWE do drop unless the adapter is changed.
Bug: webrtc:42221538
Change-Id: I49be55172aff285c55d2564ec3389f3fc7c01d62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350820
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42347}
Repeated initial probes are sent every second until
ProbeController::OnMaxAllocatedBitrate is invoked (Media is beeing sent) or 5s has passed.
Each probe has a duration of 100ms, sent in packet bursts every 20ms.
ProbeController::waiting_for_initial_probe_result_ is no longer needed
and is removed.
Setting field trial for duration between probe packets bursts are moved
from BitrateProber to ProbeController.
Bug: webrtc:14928
Change-Id: I3170533f679fc2cc2aa5402e248fa1f6996d3ccd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350640
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42323}
Replace usage of BuiltInNetworkBehaviorConfig.link_capacity_kbps in tests with DataRate link_capacity.
Bug: webrtc:14525
Change-Id: Id1c1b8d20eb2be5e9d1461704bb7c37c61c491f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42306}
This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4
Original change's description:
> Split digest methods from ssl target into digest target
>
> in an attempt to break up the monolithic ssl target.
>
> BUG=None
>
> Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#42249}
Bug: webrtc:339300437
Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42304}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
This relands the
https://webrtc-review.googlesource.com/c/src/+/349920
where patchset#1 is identical to the original change,
patchset#2 undoes (postpones) the expectation downstream propagates the Environment too.
Bug: webrtc:42220378
Change-Id: I4a9a32bb0926a875d37f3ba19dd5309e97546553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350364
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42298}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
Bug: webrtc:42220378
Change-Id: I52816628169a54b18a4405d84fee69b101f92f72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42288}
Origina description:
NetworkBehaviorInterface::RegisterDeliveryTimeChangedCallback
adds support for a network behaviour to reschedule next time DequeueDeliverablePackets should be invoked.
SimulatedNetwork::SetConfig(const BuiltInNetworkBehaviorConfig& config,
Timestamp config_update_time)
adds possibility to change the configuration at config_update_time. Delivery time of a packet currently in the narrow section, will depend on the link capacity before and after the update.
Bug: webrtc:14525
Change-Id: Idaf3a4200cfeae0683e1e1d1e98e154119ddf22e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42253}
This is a cleanup in VP9 encoder wrapper. The removed code paths were only used in tests. In prod layers are configured explicitly via VideoCodec::spatialLayers[].
Bug: webrtc:42225151
Change-Id: I1de90039488b36e3c88e788c78e675bf2ee68f9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349222
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42250}
