Rename "UpdateLayerConfig" to the more appropriate "NextFrameConfig".
Also update some comments in vp8_frame_buffer_controller.h.
Bug: None
Change-Id: Iba8227f84e33e5ebd28d2eeb10fe03e776036603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133202
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27660}
A typo in a previous CL made OnLossNotification() accept its
single argument as a const-value, rather than a const-reference.
Bug: webrtc:10501
Change-Id: I5e6f9c79f15205b75ec90a53d3fccf3dd9927e33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27659}
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).
With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.
WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@
Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
Prior to this CL, this was indicated by passing |size_bytes| = 0
to the method.
Bug: webrtc:10501
Change-Id: Icff3bb83344834dc62d62bde5ec5d05096a08e11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132712
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27620}
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
This allows the ciphersuite to be accessed from Blink code, which is
useful if we want to emit deprecation messages.
Bug: none
Change-Id: Idcf1f5402948406e4cf7c6e54b67a622a1a403ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132004
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27581}
profile-level-id for H.264 comes in through the SdpVideoFormat,
rather than through these members.
Bug: None
Change-Id: I9c4ea8873346ca16174aecf5f90a649cbaf913dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132545
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27571}
The configuration parameter filter.use_linear_filter can be used to
disable the linear filtering. Disabling the linear filter is equivalent
to runing in non-linear mode.
Bug: b/130016532
Change-Id: I8ffdf474822888b9915444bba6cc1c25ec1efe5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132552
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27566}
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Translate LossNotification RTCP messages (sequence number to
timestamp and additional information), then send the translted
message onwards to the encoder.
Bug: webrtc:10501
Change-Id: If2fd943f75c36cf813a83120318d8eefc8c595d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131950
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27545}
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.
Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
This reverts commit 7dd83e2bf73a7f1746c5ee976939bf52e19fa8be.
Reason for revert: This wasn't the cause of the break.
Original change's description:
> Revert "Refactor FrameDecryptorInterface::Decrypt to use new API."
>
> This reverts commit 642aa81f7d5cc55d5b99e2abc51327eed9d40195.
>
> Reason for revert: Speculative revert. The chromium roll is failing:
> https://ci.chromium.org/p/chromium/builders/try/linux-rel/64388
> But I can't figure out exactly what is failing, this looks suspecious.
>
> Original change's description:
> > Refactor FrameDecryptorInterface::Decrypt to use new API.
> >
> > This change refactors the FrameDecryptorInterface to use the new API. The new
> > API surface simply moves bytes_written to the return type and implements a
> > simple Status type.
> >
> > Bug: webrtc:10512
> > Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27497}
>
> TBR=brandtr@webrtc.org,steveanton@webrtc.org,solenberg@webrtc.org,ossu@webrtc.org,stefan@webrtc.org,benwright@webrtc.org
>
> Change-Id: Ia9ec70263762c34671af13f0d519e636eb8473cd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10512
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132013
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27510}
TBR=brandtr@webrtc.org,steveanton@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,ossu@webrtc.org,stefan@webrtc.org,benwright@webrtc.org
Change-Id: I8e4b7965cf1d1a1554c3b46e6245f5ad0d2dcbb4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131982
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27529}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This adds WebRTC-Vp9InterLayerPred field trial that allows to control
inter-layer prediction mode in VP9 encoder.
Bug: chromium:949536
Change-Id: Iea03db07fd21f28ab58382c5fdaac68acacc701c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131322
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27521}
Spec: https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
This already exists as a goog-stat. This CL only plumbs the value to the
new stats collector.
Note: There is currently no distinction between the extension being
missing and it being present but the value being "unspecified". Until
https://crbug.com/webrtc/10529 is fixed, this metric is only exposed if
SCREENSHARE was present.
Bug: webrtc:10452
Change-Id: Ic8723f4d0efb43ab72a560e954676facd3b90659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27520}
This reverts commit 642aa81f7d5cc55d5b99e2abc51327eed9d40195.
Reason for revert: Speculative revert. The chromium roll is failing:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/64388
But I can't figure out exactly what is failing, this looks suspecious.
Original change's description:
> Refactor FrameDecryptorInterface::Decrypt to use new API.
>
> This change refactors the FrameDecryptorInterface to use the new API. The new
> API surface simply moves bytes_written to the return type and implements a
> simple Status type.
>
> Bug: webrtc:10512
> Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27497}
TBR=brandtr@webrtc.org,steveanton@webrtc.org,solenberg@webrtc.org,ossu@webrtc.org,stefan@webrtc.org,benwright@webrtc.org
Change-Id: Ia9ec70263762c34671af13f0d519e636eb8473cd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132013
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27510}
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.
Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.
Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
This change refactors the FrameDecryptorInterface to use the new API. The new
API surface simply moves bytes_written to the return type and implements a
simple Status type.
Bug: webrtc:10512
Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27497}
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.
This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
This CL wires up the new SetRates() method of the video encoders, and
refactors a few things in the process:
Most notably, the VideoStreamEncoderInterface is update so that the
|target_headroom| parameter is replaced with |link_allocation|, meaning
that instead of indicating bitrate capacity in excess of the target
bitrate, it indicates to total network capacity allocated for the
stream including the target bitrate. This matches the VideoEncoder API.
The VideoEncoder::RateControlParameters struct gets a few new helper
methods.
In VideoStreamEncoder, instead of adding more fields to the
|last_observed_bitrate*| family, uses an optional struct that
inherits from VideoEncoder::RateControlParameters.
Bug: webrtc:10481
Change-Id: Iee3965531142ae9b964ed86c0d51db59b1cdd61c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131123
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27487}
The DefaultAudioQualityAnalyzer will read stats reports (temporarily
using the old PeerConnectionInterface::GetStats) and for each audio
stream it will collect some NetEq related stats.
When DefaultAudioQualityAnalyzer::Stop is invoked by the framework,
it will report the following metrics:
- expand_rate
- accelerate_rate
- preemptive_rate
- speech_expand_rate
- preferred_buffer_size_ms
Bug: webrtc:10138
Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27474}
This change adds FrameDecryptorInterface::Result to the FrameDecryptorInterface
API. Result contains a Status and bytes_written. This removes requiring out
parameters from the API and provides a simpler status return code for the
function. This is in response to comments suggested here:
https://webrtc-review.googlesource.com/c/src/+/131358
int FrameDecryptorInterface::Decrypt() will be removed in a follow up CL.
Bug: webrtc:10512
Change-Id: I47f19f154d1d8430acd6e4a6f433ab24c455fd51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131362
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27470}
This change makes it possible to disable AEC3's render delay
controller and delay estimator, and instead rely on an external
delay estimator. The delay is communicated via SetAudioBufferDelay.
When the feature is enabled, no echo removal will be performed
until the first delay is provided.
The delay is
Bug: b/130016532
Change-Id: I16643109d78d770ff1d2713cf247b0b9cce1bc1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131327
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27467}
This change removes the following unused parameters from the AEC3
configuration:
- render_pre_window_size_init
- render_post_window_size_init
- nonlinear_hold
- nonlinear_release
Bug: webrtc:8671
Change-Id: I8f7a3d350387cd8ada4d507c3a9fab43b7813f5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131321
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27450}
making a step for GlobalTaskQueueFactory to be optional way
to provide TaskQueueFactory
Bug: webrtc:10284
Change-Id: Ife838b3691c256820973118bc5b3cb372dea09cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130488
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27423}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
