This is to help debug a depot_tools auth problem and can be reverted
once it is solved.
Bug: skia:8394
Change-Id: I3c713fce6c6ba6edbd2498d95938b48a28eff588
Reviewed-on: https://webrtc-review.googlesource.com/101160
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Eric Boren <borenet@google.com>
Cr-Commit-Position: refs/heads/master@{#24781}
It's ok for some WebRTC dependencies not to be Chromium dependencies.
Explicitly list them instead of relying of CIDP discrimination.
Bug: chromium:855108
Change-Id: I2dafce488b28409cbce7e0c3167d92f48859084f
Reviewed-on: https://webrtc-review.googlesource.com/101000
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24774}
Until now, only revision changes were automatically rolled.
This CL detect new and removed dependencies in third_party/android_deps.
Change-Id: I4f83b7308be577115cc3ed57edd9881496428173
Bug: chromium:855108
Reviewed-on: https://webrtc-review.googlesource.com/100021
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24773}
VP9 frame rate controller is supposed to be used in screen mode only
but it was partially enabled in normal video mode. This restricts use
of VP9 frame rate controller to screen mode.
Bug: chromium:884164
Change-Id: Ie2eaa31f3364a8abccbc4171007708cf7040fc38
Reviewed-on: https://webrtc-review.googlesource.com/100424
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24769}
This makes it clearer which code actually depends on the RelayPort
class.
Bug: None
Change-Id: I7b88de1824d5b5832d2f35a8820c5c59d05441c2
Reviewed-on: https://webrtc-review.googlesource.com/100801
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24768}
When simulcast screenshare is enabled, the max bitrate for the high
quality stream can be limited based on the resolution.
This CL fixes a bug where that limit could get so low that it is below
the min bitrate of the top layer, which in turn could cause the encoder
to fail initialization.
Bug: webrtc:9761
Change-Id: I093bd0ba68fe0165e8982d169daf02cdf912c924
Reviewed-on: https://webrtc-review.googlesource.com/100682
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24767}
This is a reland of 55daf1aef65218a97eff88999e5190a2f2f6b72e.
In order to avoid problems on case insensitive file systems this CL
moves rtc_export.h to rtc_base/system (avoiding problems with build/).
Diff: https://webrtc-review.googlesource.com/c/src/+/100804/1..2.
Original change's description:
> Add RTC_EXPORT macro to export WebRTC symbols.
>
> This CL introduces the utility macro RTC_EXPORT which will let WebRTC
> developers decide which symbols are supposed to be exported/imported
> and which ones are private.
>
> RTC_EXPORT will only export/import symbols in a component build, more
> info: https://cs.chromium.org/chromium/src/docs/component_build.md.
> During a component build, the macro COMPONENT_BUILD will be globally
> defined in a consistent fashion so it is safe to rely on it to
> understand how to expand RTC_EXPORT.
> In a non component build, RTC_EXPORT will expand to nothing.
>
> Bug: webrtc:9419
> Change-Id: Ic58162783be7f5883136ade27f324d6d34fdf932
> Reviewed-on: https://webrtc-review.googlesource.com/97960
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24757}
Bug: webrtc:9419
Change-Id: Icfedea5fc3416ea1af2185de443fa879fb2dee8b
Reviewed-on: https://webrtc-review.googlesource.com/100804
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24766}
This reverts commit 55daf1aef65218a97eff88999e5190a2f2f6b72e.
Reason for revert: The build directory conflicts with the existing BUILD file on Mac where the file system is case insensitive.
Original change's description:
> Add RTC_EXPORT macro to export WebRTC symbols.
>
> This CL introduces the utility macro RTC_EXPORT which will let WebRTC
> developers decide which symbols are supposed to be exported/imported
> and which ones are private.
>
> RTC_EXPORT will only export/import symbols in a component build, more
> info: https://cs.chromium.org/chromium/src/docs/component_build.md.
> During a component build, the macro COMPONENT_BUILD will be globally
> defined in a consistent fashion so it is safe to rely on it to
> understand how to expand RTC_EXPORT.
> In a non component build, RTC_EXPORT will expand to nothing.
>
> Bug: webrtc:9419
> Change-Id: Ic58162783be7f5883136ade27f324d6d34fdf932
> Reviewed-on: https://webrtc-review.googlesource.com/97960
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24757}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,yvesg@webrtc.org
Change-Id: I9147ad010f391eeeb2e9dd0cbe7b637ebda57766
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/100803
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24763}
Add helper function to create test data,
reduce amount of unrelated details
Reduced complicated logic in tests, in particular
move most of expectation inside the tests from helpers.
Bug: None
Change-Id: I53f29a70989086c7628a0b112a45ec4567b40bf9
Reviewed-on: https://webrtc-review.googlesource.com/100380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24760}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
This CL introduces the utility macro RTC_EXPORT which will let WebRTC
developers decide which symbols are supposed to be exported/imported
and which ones are private.
RTC_EXPORT will only export/import symbols in a component build, more
info: https://cs.chromium.org/chromium/src/docs/component_build.md.
During a component build, the macro COMPONENT_BUILD will be globally
defined in a consistent fashion so it is safe to rely on it to
understand how to expand RTC_EXPORT.
In a non component build, RTC_EXPORT will expand to nothing.
Bug: webrtc:9419
Change-Id: Ic58162783be7f5883136ade27f324d6d34fdf932
Reviewed-on: https://webrtc-review.googlesource.com/97960
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24757}
A new version of RTC_EXPORT will be introduced by [1] and it will be
used by WebRTC native code.
This CL renames the current RTC_EXPORT to RTC_OBJC_EXPORT in order
to avoid to mix them. It has been decided to avoid to unify them because
RTC_OBJC_EXPORT always marks symbols with default visibility, while
RTC_EXPORT will do it only when COMPONENT_BUILD is defined.
[1] - https://webrtc-review.googlesource.com/c/src/+/97960 is
Bug: webrtc:9419
Change-Id: I56a3fc6601c72d3ad6a58f9961a00e3761dfb5da
Reviewed-on: https://webrtc-review.googlesource.com/100521
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24754}
This GN argument will be used to exclude the default implementation of
runtime_features_enabled in order to allow clients to provide a custom
implementation.
This will allow to land [1] without breaking Chromium.
[1] - https://webrtc-review.googlesource.com/c/src/+/100640
Bug: webrtc:9631
Change-Id: I4ce8ff12e277f81de42e272d8874d5bb3a4a2635
Reviewed-on: https://webrtc-review.googlesource.com/100641
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24753}
Remove function for converting uri into ExtensionType
This removes one of the lists of all supported extensions
Bug: webrtc:7472
Change-Id: I0c27239d91ef14ca4a3aa0c00588fa2b9cf10e0c
Reviewed-on: https://webrtc-review.googlesource.com/100523
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24752}
When the bandwidth estimate is volatile, and the frame rate is high,
each new frame might trigger a new video bitrate allocation that is very
close to the previous one, during BWE rampup.
This might cause unnecessarily high RTCP traffic.
This CL throttles those updates, if the allocation fullfills all of:
* Larger or the same total bitrate as the previously sent one
* Less than 10% larger bitrate compared to the previous one
* Same layers enables as the previous one
* Less than 500ms has passed since the previous one
Additionally, a call to OnEncodedImage can cause a throttled allocation
to be sent if 500ms has passed but no new call to OnBitrateUpdated has
been seen.
Bug: webrtc:9734
Change-Id: I2a17c2e512387e273e9c22bffcebd290727dc883
Reviewed-on: https://webrtc-review.googlesource.com/100560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24751}
Original CL:
https://webrtc-review.googlesource.com/c/src/+/97603
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Revert CL:
https://webrtc-review.googlesource.com/c/src/+/100305
Bug: webrtc:9535
TBR: gustaf@webrtc.org
Change-Id: I9248046b3a6a82df6221e502481836948643a991
Reviewed-on: https://webrtc-review.googlesource.com/100461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24749}
After this CL, WebRTC clients will be able to exclude the default
field_trial implementation by defining the preprocessor macro
WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT (if GN is used, it will be enough
to set rtc_exclude_field_trial_default to true).
Bug: webrtc:9631
Change-Id: I120d950e39f56f539b04ed3b921d550dd4b5ba50
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/98140
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24748}
After this CL, WebRTC clients will be able to exclude the default
metrics implementation by defining the preprocessor macro
WEBRTC_EXCLUDE_METRICS_DEFAULT (if GN is used, it will be enough
to set rtc_exclude_metrics_default to true).
Bug: webrtc:9631
Change-Id: Id6db23cc4b6c292d9f97372a8014c0c467ed0538
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/98102
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24747}
If the bandwidth is just on the edge of being able to enable a new
stream, the keyframe generated when it is enabled might be large enough
to trigger an overuse and force the stream off again.
To avoid toggling, this CL adds hysteresis so that the available
bandwidth needs to be above X% to start bitrate in order to enable the
stream. It will be shut down once available bitrate falls below the
original enabling bitrate.
For screen content, X defaults to 35.
For realtime content, X defaults to 0.
Both can be individually modified via field trials.
Bug: webrtc:9734
Change-Id: I941332d7be7f2a801d13d9202b2076d330e7df32
Reviewed-on: https://webrtc-review.googlesource.com/100308
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24745}
Also cleans up usage of the new RoundRobinPacketQueue to reduce code
bloat.
Bug: webrtc:8288
Change-Id: I90f17a4422b32c1d4e2d7d5065573157346d6a0b
Reviewed-on: https://webrtc-review.googlesource.com/100306
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24744}
Now checks the preferred channel configuration and requests implicit channel
upmixing (audio engine extends from 2 to N channels internally) if the
preferred number of channels is larger than two; i.e., initialize the
stream in stereo even if the preferred configuration is multi-channel.
To summarize: with this CL, it is now possible to use e.g. a 7.1 headset
with a native WebRTC client. All internal processing in WebRTC will be in
stereo, and the audio device will be opened up in stereo as well to match
WebRTC. Before this change, we would open up the audio device using 8
channels but that was not supported by WebRTC.
Bug: webrtc:9265
Change-Id: I1530fee28c4b8b5cda29ab6baf8d65fd391d935d
Reviewed-on: https://webrtc-review.googlesource.com/98421
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24743}
This CL:
-Adds the option to log the aec3 parameters used for a simulation.
-Cleans up the logging of the custom setting of aec3 parameters to
instead rely on the newly added logging.
Bug: webrtc:8671
Change-Id: If73a73d08e5a5077416033ded598a83fb1ade3e0
Reviewed-on: https://webrtc-review.googlesource.com/100381
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24742}
This reverts commit 958ed238603ba5a2937d28ce1c9281920d923019.
Reason for revert: Will need additional work to handle clock-drift.
Original change's description:
> AEC3: Reduce filter divergence during low-echo double-talk
>
> Bug: webrtc:9746,chromium:883264
> Change-Id: Ie3faf106fd1fd835e67d9e6794c679703af54fea
> Reviewed-on: https://webrtc-review.googlesource.com/99920
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24706}
TBR=gustaf@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9746, chromium:883264
Change-Id: Ib039eb80e2ddfc43ec52183086da2474baef65e0
Reviewed-on: https://webrtc-review.googlesource.com/100480
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24741}
The test is refitted to use the AudioProcessingStats struct to get
reference data.
The old metrics do not map entirely injectively to the new ones, so the
reference protobuf and files are updated as well.
Bug: webrtc:9535
Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36
Reviewed-on: https://webrtc-review.googlesource.com/100100
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24740}
This change enables NetEq to use the packet concealment audio (aka
PLC) produced by a decoder. The change also includes a new API to the
AudioDecoder interface, which lets the decoder implementation generate
and deliver concealment audio.
Bug: webrtc:9180
Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4
Reviewed-on: https://webrtc-review.googlesource.com/96340
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24738}
Especially for simulcast screensharing, we don't want to send constant
high bitrates of padding just to keep the bwe up since ALR probing
already handles that case.
Bug: webrtc:9734
Change-Id: I79a08fc073844628d8ad0561edd8bfcffed83fde
Reviewed-on: https://webrtc-review.googlesource.com/99120
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24734}
