This CL centralizes the render buffering in AEC3 so that all render
buffers are updated and synchronized/aligned with the render alignment
buffer.
Bug: webrtc:8597, chromium:790905
Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d
Reviewed-on: https://webrtc-review.googlesource.com/25680
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20989}
This reverts commit 8c319e951a6c59212e23af858a4c51d28b4eedc1.
Reason for revert: Increase in dropped frames and decreased send bandwidth in perf tests.
Original change's description:
> Now calculates RTT in SendSideCongestionController.
>
> Moved calculation of round trip time from transport feedback adapter to send side congestion
> controller. This reduces the role of the transport specific transport feedback adapter and
> gives more power to the congestion controller to decide how the feedback rtt should be
> calculated and used.
>
> Bug: webrtc:8415
> Change-Id: I7878d9fb32c3f4ed11993a6f39e6d9c69fab190a
> Reviewed-on: https://webrtc-review.googlesource.com/27980
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20973}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I993d00de7171a163a41b486d68b9255fd5c0f5da
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/28300
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20984}
These methods are platform specific and give access to member variables
from an unknown thread context (no thread check, no lock).
Since these methods aren't being used, it simplifies a minor refactoring project to simply delete them.
TBR=brandtr@webrtc.org
Bug: webrtc:8596
Change-Id: I85424820d171805dcc3d74317f0e51965402052a
Reviewed-on: https://webrtc-review.googlesource.com/28281
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20983}
This fixes a bug where AppRTCMobile would crash at runtime when
built without VP9 support.
Bug: webrtc:8602
Change-Id: Id2db79c3ff8136f06dc049afcc5197e9356fd25b
Reviewed-on: https://webrtc-review.googlesource.com/27983
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20982}
I ran into this while looking at the rtc::Event class.
TBR=mbonadei@webrtc.org
Bug: webrtc:8596
Change-Id: Iee00b15381548b69f6cb9485788c787bde23494a
Reviewed-on: https://webrtc-review.googlesource.com/28280
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20981}
Notifications are printed for gaps in seq number, capture timestamp, arrival and send times for RTP and RTCP, and high average loss.
The notifications are printed to stderr by default, but internally they are represented as subclasses to a TriageNotification base class in order to facilitate other output formats.
Initially, this is only run if the event_log_visualizer is given the flag --print_triage_notifications.
Only the first (LOG_START, LOG_END) segment is processed.
Bug: webrtc:8383
Change-Id: If43ef7f115f622fa5552dc50951a11d5f9e3cbaa
Reviewed-on: https://webrtc-review.googlesource.com/8720
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20974}
Moved calculation of round trip time from transport feedback adapter to send side congestion
controller. This reduces the role of the transport specific transport feedback adapter and
gives more power to the congestion controller to decide how the feedback rtt should be
calculated and used.
Bug: webrtc:8415
Change-Id: I7878d9fb32c3f4ed11993a6f39e6d9c69fab190a
Reviewed-on: https://webrtc-review.googlesource.com/27980
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20973}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Idf275495910f651ec35f641611926e62414daa9a
Reviewed-on: https://webrtc-review.googlesource.com/23610
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20972}
Header files base/videosinkinterface.h and base/videosourceinterface.h
were not part of any target (because they cause 2 dependency cycles).
This CL uncomment them so GN can keep dependencies under control, the
2 dependency cycles will be removed as part of webrtc:6828.
Bug: webrtc:6828
Change-Id: I5c5580facc010ba619e105a9b8a572ac70169a01
Reviewed-on: https://webrtc-review.googlesource.com/27621
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20970}
The data-channel only target "peerconnectionfactory_no_media_objc"
transitively depends on libvpx via "peerconnectionfactory_base_objc".
This CL breaks the dependency between "objc_peeerconnectionfactory_base"
and libvpx moving RTCVideoCodecVP8.mm and RTCVideoCodecVP9.mm to
"peerconnectionfactory_objc" (together with RTCVideoCodecH264.mm).
Bug: webrtc:8594
Change-Id: Idfe3024163012925f017ad8c585b7ae21e86c319
Reviewed-on: https://webrtc-review.googlesource.com/27480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20969}
When enabling SDES in BaseChannel, SrtpTransport::EnableExternalAuth
should be called if the external authenication is enabled which is
not covered by the current test set.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ibbf458516a521a488e8e3bb4a5a29fca70a627f5
Reviewed-on: https://webrtc-review.googlesource.com/27761
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20960}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=magjed@webrtc.org
Bug: None
Change-Id: I78842b6bb8ae345bcb852feee3908fdaf955c664
Reviewed-on: https://webrtc-review.googlesource.com/23574
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20956}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=sprang@webrtc.org
Bug: None
Change-Id: Ic429f28a8610ca798e29c45ec1f64604d6f9687f
Reviewed-on: https://webrtc-review.googlesource.com/23603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20955}
Add a method to DtlsSrtpTransport to cache the RTP Absolute Send Time
extension id. The method would be called when using DTLS-SRTP with
external authentication.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ie9edb9382cbb4cf43eea5da3030991a0d20293a5
Reviewed-on: https://webrtc-review.googlesource.com/27260
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20947}
This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This was only used to ensure compatibility with external projects.
Now that everyone has moved to RtpTransceiverDirection, these
methods are redundant.
Bug: webrtc:8558
Change-Id: Iff5a8d13f9a4300d06902fa0441ceaeebf6809a2
Reviewed-on: https://webrtc-review.googlesource.com/24746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20945}
