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}
This enables us to migrate a internal project from depending on implementation
details of our LOG macros.
Bug: webrtc:9278
Change-Id: I41408aec40e26d2b55df58360f94f5ceaa57af4e
Reviewed-on: https://webrtc-review.googlesource.com/98001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24597}
This is a reland of 711a31aead9007e42dd73c302c8ec40f9e931619
Changes since original landing:
Rename methods only used by tests, mainly via FakeClock,
MessageQueueManager::ProcessAllMessageQueues
--> ProcessAllMessageQueuesForTesting
MessageQueue::IsProcessingMessages
--> IsProcessingMessagesForTesting
Fix the handling of null rtc::Thread::Current() in
ProcessAllMessageQueuesInternal().
Add override Thread::IsProcessingMessagesForTesting() to return false
for the wrapped main thread, unless it's also the current thread. In
tests, the main thread is typically not processing any messages,
but blocked in an Event::Wait().
Original change's description:
> Explicitly wrap main thread in test_main.cc.
>
> Bug: webrtc:9714
> Change-Id: I6ee234f9a0b88b3656a683f2455c3e4b2acf0d54
> Reviewed-on: https://webrtc-review.googlesource.com/97683
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24560}
Bug: webrtc:9714
Change-Id: I6f022d46aaf1e28f86f09f2d68c1803b69770126
Reviewed-on: https://webrtc-review.googlesource.com/98060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24596}
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}
This reverts commit bb19276a325a5f9fce4afa245aa14ec2a4b1a41d.
Reason for revert: breaks downstream project
Original change's description:
> Use AsyncInvoker in PeerConnection instead of MessageHandler
>
> Bug: webrtc:9702
> Change-Id: I89d66d1165a096601aed37b8febad60620073899
> Reviewed-on: https://webrtc-review.googlesource.com/97180
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24515}
TBR=steveanton@webrtc.org,shampson@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9702
Change-Id: Ibfe507cd1593f7000e11f9a17313a016307381cb
Reviewed-on: https://webrtc-review.googlesource.com/98302
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24591}
Fixes issues when using the display arrangement utility. The region
outside of the screens had random data when re-arranging displays
while capturing. This CL makes sure this region is always black.
Bug: webrtc:9703
Change-Id: I1481dd0f1b4584e75926755f9b8a6e5161cd5904
Reviewed-on: https://webrtc-review.googlesource.com/97184
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24589}
As downside it disallows to destroy RtcpTransceiver on the TaskQueue
without prio call to the Stop function
BUG: webrtc:8239
Change-Id: I236b9aff7a0746044dd764c619174cc5ac03d27f
Reviewed-on: https://webrtc-review.googlesource.com/98120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24587}
Also enables support for all hardware implementations. Renames
HardwareVideoDecoderFactory to MediaCodecVideoDecoderFactory. Renames
HardwareVideoDecoder to AndroidVideoDecoder.
Bug: webrtc:8538
Change-Id: I9b351f387526af4da61fb07c07fb4285bd833e19
Reviewed-on: https://webrtc-review.googlesource.com/97680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24586}
This is a reland of 9bb55fc09b6bfa00cba7779c37ad6c39b4206f7a
Original change's description:
> Reland "Update video_quality_analysis to align videos instead of using barcodes"
>
> This is a reland of d65e143801a7aaa9affdb939ea836aec1955cdcc
>
> The binary for frame_analyzer.cpp is precompiled and stored in the cloud, so it
> won't automatically pick up change to the source file. Therefore, restore all
> old code to be backwards compatible.
>
> Original change's description:
> > Update video_quality_analysis to align videos instead of using barcodes
> >
> > This CL is a follow-up to the previous CL
> > https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> > logic for aligning videos. This will allow us to easily extend
> > video_quality_analysis with new sophisticated video quality metrics.
> > Also, we can use any kind of video that does not necessarily need to
> > contain bar codes. Removing the need to decode barcodes also leads to a
> > big speedup for the tests.
> >
> > Bug: webrtc:9642
> > Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> > Reviewed-on: https://webrtc-review.googlesource.com/94845
> > Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24423}
>
> TBR=phensman@webrtc.org,phoglund@webrtc.org
>
> Bug: webrtc:9642
> Change-Id: Id8d129ce103284504c67690f8363c03eaae3eee7
> Reviewed-on: https://webrtc-review.googlesource.com/96000
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24429}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: Ic248b7831ae148251a1a4ebeec5d154286f91a0a
Reviewed-on: https://webrtc-review.googlesource.com/98080
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24583}
This feature went to stable with M69. Switch is in M69 and M70 banches.
Since tot is now M71 and we have not seen any issues, let's clean this
up.
Bug: webrtc:9634
Change-Id: I708bab55b0443d0873b09dd5b71cdfad72397a7a
Reviewed-on: https://webrtc-review.googlesource.com/98002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24581}
This is a reland of b2c0e8f60fad10e2786e5e131136a0da1299d883
Original change's description:
> Add tool for aliging video files
>
> This class adds logic for aligning a test video to a reference video
> by an algorithm that maximizes SSIM between them. Aligned videos will be
> easier to run video quality metrics on. This is a generic way of
> aligning videos and can be replace the intrusive barcode stamping that
> we currently use. This will be done in a follow-up CL.
>
> Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c
> Bug: webrtc:9642
> Reviewed-on: https://webrtc-review.googlesource.com/94773
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24407}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: I35d6b0e598335b8d80fbfa37ba06d5c651bda4f6
Reviewed-on: https://webrtc-review.googlesource.com/98040
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24580}
This problem shouldn't be severe enough to require a crash, and it can be triggered by untrusted network data.
We'll print a warning instead.
Bug: chromium:877790
Change-Id: Ie1f33d85dc7f0ccbbc1d027b73f77b964c6f7e46
Reviewed-on: https://webrtc-review.googlesource.com/96880
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24579}
Second round of the new Windows ADM is now ready for review. Main
changes are:
Supports internal (automatic) restart of audio streams when an active
audio stream disconnects (happens when a device is removed).
Adds support for IAudioClient3 and IAudioClient2 for platforms which
supports it (>Win8 and >Win10).
Modifies the threading model to support restart "from the inside" on
the native audio thread.
Adds two new test methods for the ADM to emulate restart events or
stream-switch events.
Adds two new test methods to support rate conversion to ensure that
audio can be tested in loopback even if devices runs at different
sample rates.
Added initial components for low-latency support. Verified that it works
but disabled it with a flag for now.
Bug: webrtc:9265
Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a
Reviewed-on: https://webrtc-review.googlesource.com/86020
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24578}
Limited range seems to be more used than full range and many Android
components already use limited range. This includes FileVideoCapturer,
VideoFileRenderer and HW codecs.
Bug: webrtc:9638
Change-Id: Iadd9b2f19020c6a25bde5e43a28e26a6230dde42
Reviewed-on: https://webrtc-review.googlesource.com/94543
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24576}
This allow to destroy the RtcpTransceiverImpl off the task queue
with assumption it is destroyed after task queue.
i.e. it allows to post destruction of RtcpTransceiverImpl to the TaskQueue without waiting.
BUG: webrtc:8239
Change-Id: I4bea7a6d2edc97061ebd00f2f275c1ab827bc3c5
Reviewed-on: https://webrtc-review.googlesource.com/97160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24574}
