Also read and apply settings when parsing and replaying dumps.
The implementation contains
* an extra field in debug.proto for the runtime settings
* code in AudioProcessingImpl to initiate the logging of the RS to the
AecDump
* code in aec_dump/ to log the RS in the AecDump
* code in test/ for re-playing the RS. E.g. for APM simulation with
audioproc_f.
Bug: webrtc:9138
Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f
Reviewed-on: https://webrtc-review.googlesource.com/70502
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24647}
This reverts commit ae9e188e67a489db597224e3cfcfdee04edf0cba.
Reason for revert: Verify if this causes chromium:882358.
Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}
TBR=sprang@webrtc.org,ssilkin@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9682, chromium:882358
Change-Id: Idc4051eef72104823038ed9139bb9c75018f7d86
Reviewed-on: https://webrtc-review.googlesource.com/99082
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24646}
Today, the internal frame dropper in libvpx vp8 encoder is enabled or
disabled based on video or screen content. This is then expected to
match up with screenshare vs default temporal layers implementation.
This cl makes libvpx query the temporal layers implementation as well,
breaking this implicit dependency and allows frames to be dropped if
default temporal layers is used with screen content.
Bug: webrtc:9734
Change-Id: If2523a211f4929f16e65a02fa7a6b4edf7328571
Reviewed-on: https://webrtc-review.googlesource.com/99062
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24645}
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.
Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
Improve HasAttribute to handle the case where the beginning of an
attribute name is also an attribute name in it self. Two attributes
that have this relation are extmap-allow-mixed and extmap.
Bug: webrtc:9712
Change-Id: Iee660cc6e3dc7f2e7c56664a4f0ffb298eca9208
Reviewed-on: https://webrtc-review.googlesource.com/97422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24640}
The tests crashed and were disabled temporarily. The crash was probably
caused by chromium:879307 which was fixed recently.
Bug: webrtc:9506
Change-Id: I08872c0370c9cf5dc4769daf68b7c61135a55c9e
Reviewed-on: https://webrtc-review.googlesource.com/99080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24638}
Currently, invalid PeerConnection object is returned. With this change,
null is returned instead. This can be more easily handled in the
application layer.
Bug: webrtc:9440
Change-Id: I44dfee81a681f033b8d336c999d43ff1c69fb015
Reviewed-on: https://webrtc-review.googlesource.com/98480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24637}
Non trivially destructible objects with static storage are disallowed
by the style guide.
This CL just removes 'static' since these objects are constructed once
or twice in the entire application.
Bug: webrtc:9693
Change-Id: I7509e2c088dd5ec0ac13f08053ecb76cf8259d90
Reviewed-on: https://webrtc-review.googlesource.com/98840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24634}
At cost of removing assumption callbacks can't be used after destructor.
Bug: webrtc:8239
Change-Id: Id79f7553528cf6c102d3ee0bf7aa2de5b0437d2a
Reviewed-on: https://webrtc-review.googlesource.com/98860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24632}
Currently we check if a message should be printed at the call site using LogMessage::Loggable, in the LogMessage itself using LogMessage::IsNoop and in LogMessage::OutputToDebug using log_to_stderr_.
This change unifies the first two of these into a early return in Log().
Bug: webrtc:8982
Change-Id: I462b1cf63c44fec46e5c59b147b2b99605aaae0c
Reviewed-on: https://webrtc-review.googlesource.com/98820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24630}
Rely on C++11 thread-safe initialization on first call to
MessageQueueManager::Instance(), in the same way as for
ThreadManager::Instance().
Bug: None
Change-Id: I26244f90c5d7f94a2454688297f55bf96617e78c
Reviewed-on: https://webrtc-review.googlesource.com/97721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24625}
Assume that stream has single temporal layer if number of frames in GOF
is set to zero (valid case).
Bug: chromium:879584
Change-Id: I7ced082190e40c1bf4cc1468babfd98b0a61f0dd
Reviewed-on: https://webrtc-review.googlesource.com/98800
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24622}
Using signed integers allows to centralize checking of edge cases
in RtpPacketizer::SplitAboutEqually and
reduce chance of hitting issues with size_t underflow
Bug: webrtc:9680
Change-Id: Ic05bf0a9565a277c4608f43061ca46cf44e82d08
Reviewed-on: https://webrtc-review.googlesource.com/98602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24618}
This CL performs some cleanups on multiplex files:
- Adds more comments to factory about usage.
- Moves image packer outside /include as it doesn't need to be public.
- Other small lint issues.
Bug: webrtc:9632
Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32
Reviewed-on: https://webrtc-review.googlesource.com/98700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24615}
This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.
To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.
Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
This change allows an implementer to provide a custom error code to be returned
on failure with 0 being reserved for success. It also modifies the output size
APIs to be more specific and requests the number of bytes written to be returned
so we can determine how many bytes we need to send.
Bug: webrtc:9681
Change-Id: I13d34861bf851527fcbb350d0cfb480c0f95a6b3
Reviewed-on: https://webrtc-review.googlesource.com/98720
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24613}
This is experimental interface for media transport.
The goal is to refactor WebRTC codebase to send/receive frames via media transport interface. It will allow us to have different media transport implementations in the future, including QUIC-based media transport.
Bug: webrtc:9719
Change-Id: I64e0b69d18c212e1ed0a08c6904578c3dfbe3af7
Reviewed-on: https://webrtc-review.googlesource.com/95960
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24612}
This changes adds the API surface for injecting the FrameEncryptor and FrameDecryptor from Java.
This assumes that the API User will be able to provide native implementations of both the Encryptor
and Decryptor. Optional Java implementations may come later but due to the significant performance
issues around copying every frame across the JNI boundary it doesn't seem like a good idea to support
a non native backed implementation for now.
Bug: webrtc:9681
Change-Id: Ib4471e69fdf0a99705f824de652c621637b92326
Reviewed-on: https://webrtc-review.googlesource.com/96865
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24610}
Using the mac framework bundle's path should be more correct than the
hardcoded version used previously.
Bug: webrtc:9627
Change-Id: I706c7961f21ba921503ed68b6e79ea0838e664d5
Reviewed-on: https://webrtc-review.googlesource.com/98620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24606}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.
It also adds the possibility to suppress these warnings because
they trigger in a few places.
The long term goal is to avoid regressions on this and remove all the
suppressions.
Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
Bug: webrtc:3380
Change-Id: I39c6b35d24182475b33a7a321cdf3b3ac9b8979a
Reviewed-on: https://webrtc-review.googlesource.com/97861
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24602}
This restricts the maximum burst length to receive times 100 ms apart.
Currently packets will be considered part of the same burst as long as
they are received within 5 ms from the previous packet. This can happen
when recovering from large network buffers.
Bug: webrtc:9718
Change-Id: I73027ddaae922cb7bb9a477cf29b4f0036ce6966
Reviewed-on: https://webrtc-review.googlesource.com/98280
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24601}
Since the checks build target is included pretty much everywhere it's hard to reuse any of our other string handling tools. Instead I largely copied AppendFormat from StringBuilder.
Bug: webrtc:8982
Change-Id: I7b23d65d5c21a784113f74d9cbbfded5a616cfc4
Reviewed-on: https://webrtc-review.googlesource.com/97423
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24600}
