This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.
Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.
Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24917}
In the case a frame_object is kept for some time before it is deleted,
it may happend that a new frame is received with overlapping sequence
numbers. If the old frame_object is removed while receiving the new
frame there used to be a crash.
Bug: webrtc:9629
Change-Id: I270a8caa2b58b73c000542aa504c0ebe277d49c4
Reviewed-on: https://webrtc-review.googlesource.com/102683
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24914}
This CL removes kNumMicrosecsPerSec and kNumMillisecsPerSec from
modules/audio_device/win/core_audio_utility_win.h.
kNumMillisecsPerSec was unused, while kNumMicrosecsPerSec has been
replaced by rtc::kNumMicrosecsPerSec.
Bug: webrtc:9693
Change-Id: I560aa9dad2bfb94a9bf67d3b9941700f1948086b
Reviewed-on: https://webrtc-review.googlesource.com/102860
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24906}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.
Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
In the effort of enabling -Wglobal-constructors and
-Wexit-time-destructors, WebRTC has to remove the Winsock global
initializer.
This will also remove it from Chromium (since it was unused).
After this CL, applications will have to explicitly initialize Winsock
before using WebRTC, this can be done by using the class
rtc::WinsockInitializer provided in rtc_base/win32socketinit.h.
Bug: webrtc:9693, webrtc:9754
Change-Id: I4aae12ff43671ef2713a6fc4592e20759dc6b495
Reviewed-on: https://webrtc-review.googlesource.com/99660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24903}
This is a reland of e307d56bd7e192c354871a739bc0133d88cb5379
options.yuv_directory would be unset if vmaf was not used.
It now gets set to None.
Also adds a try-finally around the temp directory for YUV files.
Original change's description:
> Add option to call VMAF in compare_videos.py.
>
> VMAF compares videos on several metrics and produces a unified score.
>
> Calling it from compare_videos required passing in a path to a VMAF
> executable and a model.
>
> VMAF needs to compare aligned videos in YUV format, so two videos
> (ref and test) will be saved by frame_analyzer after it has aligned
> them.
>
> Bug: webrtc:9642
> Change-Id: Idddfcf6b1b235e7f925696ffc38938fb84c4ff9e
> Reviewed-on: https://webrtc-review.googlesource.com/102140
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24876}
Bug: webrtc:9642
Change-Id: I1d04a56090e68df47dc3e6b7e710384244470d0c
TBR: phoglund
Reviewed-on: https://webrtc-review.googlesource.com/102544
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24896}
This reverts commit 2fbb83b16b4c2c1712cbe898ca3ba42d6da3e96f.
Reason for revert: Speculative revert over failing Chromium bot:
https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28M%20Nexus5X%29/117
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I1f8a27ac291f2cdc16c8daa32e399b74d489dbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/102642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24895}
This makes it easier to debug issues related to double dispose /
use after dispose.
Bug: webrtc:7566, webrtc:8297
Change-Id: I07429b2b794deabb62b5f3ea1cf92eea6f66a149
Reviewed-on: https://webrtc-review.googlesource.com/102540
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24894}
These submodules implicitly rely on low cut filtering being enabled.
This CL clarifies a distinction:
High pass filtering is a feature that users can enable, according to the WebRTC standard.
Low cut filtering is a processing effect that is applied when any of the following is active:
- high pass filter
- noise suppression
- builtin echo cancellation
Bug: webrtc:9535
Change-Id: I9474276fb11354ea3b01e65a0699f6c29263770b
Reviewed-on: https://webrtc-review.googlesource.com/102600
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24892}
Adds SimulatedTimeClient, a class that simulates time so congestion
controllers can be tested using the Scenario test framework without
running in real time.
This allows using simplified scenario tests as unit tests, narrowing
the gap between end to end tests and unit tests.
Bug: webrtc:9510
Change-Id: I61ab388bd610f636b926675b1f14b8d85e3c1114
Reviewed-on: https://webrtc-review.googlesource.com/99801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24890}
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.
Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
This is a sibling CL to a similar one for EchoCancellation:
https://webrtc-review.googlesource.com/c/src/+/97603
- EchoControlMobileImpl will no longer inherit EchoControlMobile.
- Removes usage of AudioProcessing::echo_control_mobile() inside most of
the audio processing module and unit tests.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (comfort noise, routing mode), but prints an
error message when unsupported settings are encountered.
Tested: audioproc_f with .wav and aecdump inputs.
Bug: webrtc:9535
Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
Reviewed-on: https://webrtc-review.googlesource.com/101621
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24888}
Symbols under sdk/ are now exported using RTC_OBJC_EXPORT, while
RTC_EXPORT is used for C++ symbols.
Bug: webrtc:9419
Change-Id: Icdf7ee0e7b3faf4d7fec33e9b33a3b13260f45b7
Reviewed-on: https://webrtc-review.googlesource.com/102461
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24886}
There were remnants of use in proxy_unittest.cc, instantiating an
HttpListenServer but not using it for anything.
Also trim down httpcommon.h, the only function still in use is
HttpAuthenticate.
Bug: webrtc:6424
Change-Id: I9b122dedd6e8c923ed7bc721a336fe54192328c4
Reviewed-on: https://webrtc-review.googlesource.com/102141
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24884}
This reverts commit e307d56bd7e192c354871a739bc0133d88cb5379.
Reason for revert:
Breaks client.webrtc.perf bots. Example failure:
https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20(L%20Nexus7.2)/8635
AttributeError: Values instance has no attribute 'yuv_directory'
Original change's description:
> Add option to call VMAF in compare_videos.py.
>
> VMAF compares videos on several metrics and produces a unified score.
>
> Calling it from compare_videos required passing in a path to a VMAF
> directory, where there should be a C++ wrapper executable and a model.
> For now, the relative paths to those are constant.
>
> VMAF needs to compare aligned videos in YUV format, so two videos
> (ref and test) will be saved by frame_analyzer after it has aligned
> them.
>
> Bug: webrtc:9642
> Change-Id: Idddfcf6b1b235e7f925696ffc38938fb84c4ff9e
> Reviewed-on: https://webrtc-review.googlesource.com/102140
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24876}
TBR=phoglund@webrtc.org,sakal@webrtc.org,phensman@webrtc.org
Change-Id: I3e1dc98d7dfc0309ee2934cb3a978eecf274c477
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/102561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24883}
This is a reland of 529d0d9795b81dbed5e4231f15d3752a5fc0df32
Original change's description:
> Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
>
> Preparation for deleting EnableFrameRecordning, and also a step
> towards landing of the new VideoStreamDecoder.
>
> Bug: webrtc:9106
> Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> Reviewed-on: https://webrtc-review.googlesource.com/97660
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24861}
Bug: webrtc:9106
Change-Id: I2eb894773b3f33ff6a980e8008e8248607e32668
Reviewed-on: https://webrtc-review.googlesource.com/102480
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24882}
These changes are based on finding when using Tsan v2. More changes are
needed before usage of the THREAD_SANITIZER build flag can be removed.
Hence, all tests are still ignored when this flag is set. The changes
are still improvements.
See https://bugs.chromium.org/p/webrtc/issues/detail?id=9778#c10
for more details.
Bug: webrtc:9778
Change-Id: I1266cec48165046dcffc16f104ec5b88b41500b2
Reviewed-on: https://webrtc-review.googlesource.com/102440
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24880}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
This CL removes killswitches for code that has been properly tested in
experiments and is to be considered to be permanent.
The changes have been tested for bitexactness.
Bug: webrtc:8671
Change-Id: I0f9db16f377390d9dd3779096da91f3abc0fb4a5
Reviewed-on: https://webrtc-review.googlesource.com/102360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24877}
VMAF compares videos on several metrics and produces a unified score.
Calling it from compare_videos required passing in a path to a VMAF
directory, where there should be a C++ wrapper executable and a model.
For now, the relative paths to those are constant.
VMAF needs to compare aligned videos in YUV format, so two videos
(ref and test) will be saved by frame_analyzer after it has aligned
them.
Bug: webrtc:9642
Change-Id: Idddfcf6b1b235e7f925696ffc38938fb84c4ff9e
Reviewed-on: https://webrtc-review.googlesource.com/102140
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24876}
