Storing raw frames in memory leads to the application running out of
memory during long running tests.
Bug: b/80409365
Change-Id: I9fea171dc76cf0b3b6bba64c60a91353f69fafaa
Reviewed-on: https://webrtc-review.googlesource.com/79581
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23430}
A previous change caused a regression in the congestion window behavior.
This CL restores previous behavior.
Bug: webrtc:8415
Change-Id: Id2e42d66bcfb58780c98da2227da39b970f26f0e
Reviewed-on: https://webrtc-review.googlesource.com/79483
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23428}
This removes usage of the old OnFailure methods on CreateSessionDescriptionObserver
and SetSessionDescriptionObserver, so that WebRTC will continue to compile
once all the default implementations are removed.
Bug: chromium:589455
Change-Id: Id67295b3ad0c30d24d79589c2041acdd507a19f3
Reviewed-on: https://webrtc-review.googlesource.com/78480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23427}
bugs.webrtc.org/8439 introduces application data that could e.g. contain
timestamps. We would like to take different actions for this data
depending on whether this is the first time a packet is being sent.
Bug: webrtc:8906
Change-Id: Ib370d76beec2960d961bf44391930faa4b193479
Reviewed-on: https://webrtc-review.googlesource.com/77643
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Petter Strandmark <strandmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23426}
It should be the responsibility of the fixture user to provide the exact
codecs that should be tested instead. This reduces the coupling between
the test fixture and the codec instantiation.
Bug: webrtc:9317
Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b
Reviewed-on: https://webrtc-review.googlesource.com/79147
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23425}
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I7b7e4769772d67cc5112969fefd4e56c6c72432e
Reviewed-on: https://webrtc-review.googlesource.com/76600
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23419}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
When CongestionWindowPushback experiment is enabled, the pacer is oblivious to the congestion window. The relation between outstanding data and the congestion window affects encoder allocations directly.
Bug: None
Change-Id: Iaacc1d460d44a4ff2d586934c4f9ceb067109337
Reviewed-on: https://webrtc-review.googlesource.com/74922
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23411}
In flexible mode, use VP9E_GET_SVC_REF_FRAME_CONFIG to get indices of
reference frame buffers and buffers update by encoded frame.
Set inter_pic_predicted to true only if encoder actually used temporal
prediction.
Bug: webrtc:9244, webrtc:9270
Change-Id: I4e439abeab9e063d50abdcefc59bf58d6596ea6c
Reviewed-on: https://webrtc-review.googlesource.com/74780
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/master@{#23410}
Also deletes api/videosinkinterface.h, which was moved to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I01211c2862f964196f8e45155cbbb7f4f0204483
Reviewed-on: https://webrtc-review.googlesource.com/76420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23408}
Only remaining user was rtc_base/byteorder.h, which is changed to use
rtc_base/system/arch.h.
Bug: webrtc:6853
Change-Id: If3b21831adc60adfd989061027d661867c938a0f
Reviewed-on: https://webrtc-review.googlesource.com/78740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23406}
Thus we don't need to initialize new members added to the structure
in the future.
Bug: None
Change-Id: Id9f5b127c224660f3016973261045b4231a617c1
Reviewed-on: https://webrtc-review.googlesource.com/79080
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23404}
This reverts commit 34b1bc72995d4aa52f2e96b6c60a6ec9eacbde48.
Reason for revert: The issue in libvpx has been fixed.
Original change's description:
> Disable flaky test: FullStackTest.VP9SVC_3SL_High
>
> Following a change in libvpx, FullStackTest.VP9SVC_3SL_High has
> become flaky. It will be disabled until the libvpx issue is fixed.
>
> Bug: webrtc:9293
> NOTRY: true
> Change-Id: Ib375363bdefdbb4104130a1f0f02ea34dc26e7f9
> Reviewed-on: https://webrtc-review.googlesource.com/77663
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23319}
TBR=eladalon@webrtc.org,sprang@webrtc.org,ssilkin@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9293
Change-Id: I80f33baac35a1fc8d446a7639fa64a94774dde4a
Reviewed-on: https://webrtc-review.googlesource.com/78900
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23402}
In this work, we change the behavior of the gain limiter so it also looks at the energy
on farend around the default delay for deciding the suppression gain
that should be applied at the initial portion of the call.
Bug: webrtc:9311,chromium:846724
Change-Id: I0b777cedbbd7fd689e72070f72237296ce120d3c
Reviewed-on: https://webrtc-review.googlesource.com/78960
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23400}
This saves having to iterate trough all packets in flight to compute the
number of outstanding bytes.
Bug: webrtc:8415
Change-Id: I35b135f37649a38b44a36d300af42a815f85192d
Reviewed-on: https://webrtc-review.googlesource.com/77727
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23398}
All decoders are injectable, no need to create built-in codecs from
there.
Bug: webrtc:7925
Change-Id: Iabf3d59a8e4d721ad29386acbf138b7e5992ce5e
Reviewed-on: https://webrtc-review.googlesource.com/72441
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23397}
Letting the delay estimator operate at a sampling frequency of 2 kHz
with audio between 0 and 1 kHz makes it sensitive to noisy environments.
This CL bandpass filters the 16 kHz signal before downsampling to 2 kHz
in a way that the downsampled 2 kHz signal contains audio between 1 and
2 kHz. It also sets downsampling factor 8 as default which significantly
reduces computational complexity.
Bug: webrtc:9288,chromium:846615
Change-Id: Iaf67898a1a14326cd61bb7f81c14d3c12a697c8d
Reviewed-on: https://webrtc-review.googlesource.com/78703
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23395}
It's always been there, and there's no security risk.
Bug: chromium:843477
Change-Id: I6121943f23b477300cf60ffc4858ef0ab43466dc
Reviewed-on: https://webrtc-review.googlesource.com/78782
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23393}
It was being parsed, but not serialized. Meaning that if you set a
remote description with a=ice-lite, and then read the remoteDescription
attribute, it doesn't contain a=ice-lite.
NOTRY=True
Bug: webrtc:6668
Change-Id: Ia3c56d876c317b5af71a1f383f238d1e86f06a01
Reviewed-on: https://webrtc-review.googlesource.com/78821
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23391}
This is needed for downstream users of the impl, as we currently pull
in Chromium specifics in the android_codec_factory_helper. Further,
the downstream users should explicitly supply their own factories
if they do not want to use the internal ones.
Bug: None
Change-Id: Ia7b01a66aadaba3d5accf44e5ca38e1a319e4e34
Reviewed-on: https://webrtc-review.googlesource.com/78420
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23390}
Extra switches to GN could be passed via --extra-gn-switches.
Extra switches to Ninja could be passed via --extra-ninja-switches.
They could be used in different scenarios, when additional switches
need to be passed to GN or Ninja. For example, when diagnosing
build issues extra switch `-v` could be passed to enable
verbose logging of GN and Ninja.
Bug: None
Change-Id: I09d18a57b3df4e698784fb7d58c02e8adecddefa
Reviewed-on: https://webrtc-review.googlesource.com/78722
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23388}
This class receives data about video frames from ReceiveStatisticsProxy,
calculates spatial and temporal quality metrics and outputs them to UMA
stats. It is all done in a separate class because it will be further
extended to calculate aggregated quality metrics in the future.
Bug: webrtc:9295
Change-Id: Ie36db83e10c0e8da0b9baa392651cb9a67a54a80
Reviewed-on: https://webrtc-review.googlesource.com/78220
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23387}
This CL replaces components in the congestion controller module
that are identical to equivalent components in the rtp and goog_cc
subfolder. Some redundant components are left as they were not
trivial to replace.
Bug: webrtc:8415
Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37
Reviewed-on: https://webrtc-review.googlesource.com/78521
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23384}
