This class will be used for filtering remote clock offset in rtp streams.
It is a separate wrapper around PercentileFilter because it will be used
in that form in several places.
Bug: webrtc:8468
Change-Id: If1f6c38ac1ffa02232c1aed5512b92878b1c346a
Reviewed-on: https://webrtc-review.googlesource.com/17841
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20531}
At the beginning of the call, when rtt is not yet estimated, SR packets
are not used for estimation. Yet, it may happen that on some non-SR
RTCP packet RTT would become available. At that time an old SR will be
used for remote clock estimation. This will lead to remote clock offset
to the past too much.
Bug: webrtc:8468
Change-Id: I1bdbd56a7bab1c28e73987e5fb307f8e7382b045
Reviewed-on: https://webrtc-review.googlesource.com/16840
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20528}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
Using layoutSubviews is a simpler solution to achieveing the desired
effect. Plus this will get rid of warrnings on iOS 11
Bug: webrtc:8467
Change-Id: Idaa041b7a0ed889905d97f645408fb3437154e73
Reviewed-on: https://webrtc-review.googlesource.com/17380
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20523}
WebRTC rolls into Chromium are failing, we should fix it ASAP.
Log:
FAILED:
obj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe
"e:\b\c\win_toolchain\vs_files\88c3b62e1eb0893b8cd57e3f4859c3af27907f64\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe"
/nologo /showIncludes
@obj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj.rsp
/c
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
/Foobj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj
/Fd"obj/third_party/webrtc/modules/video_coding/webrtc_stereo_cc.pdb"
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc(134):
error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc(134):
warning C4267: 'argument': conversion from 'size_t' to 'uint32_t',
possible loss of data
Bug: chromium:780411
Change-Id: Ia80f4551d0efeebc6d084e951f5c25e8b9401250
Reviewed-on: https://webrtc-review.googlesource.com/17781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20522}
There is a crash happening in this neighbourhood, so adding
CHECKs to tease it out explicitly.
BUG=webrtc:8481
Change-Id: I79a2ec8fd838f4a4735a04496e363b72975919ec
Reviewed-on: https://webrtc-review.googlesource.com/17361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20520}
This reverts commit 90bace095806a635411edd40fb8490a144e59e63.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
This reverts commit 54e41dd08a06f978a6d123f19f5bbe39370f04ea.
Reason for revert: We are reverting also https://webrtc-review.googlesource.com/c/src/+/16180, so this CL will be included in the re-land of https://webrtc-review.googlesource.com/c/src/+/16180.
Original change's description:
> Remove const from ThreadChecker in NullAudioPoller.
>
> TBR=henrika@webrtc.org,solenberg@webrtc.org
>
> Bug: webrtc:8482
> Change-Id: Ib2738224e776618c692db95cd9473335bc17be15
> Reviewed-on: https://webrtc-review.googlesource.com/17540
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20505}
TBR=terelius@webrtc.org
Change-Id: I27c70ce331043ffdfec676c7e1a51e741d2fe770
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8482
Reviewed-on: https://webrtc-review.googlesource.com/17700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20511}
This, presumably, will make LUCI tryjobs run for people
opted into the experiment.
Bug: chromium:749455,chromium:776347
Change-Id: I25e3a011810aa47e5f245e1eb0ea2547e29fa3ff
Reviewed-on: https://webrtc-review.googlesource.com/16420
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20506}
uint8_t was being printed as a char; a conversion to int was necessary.
Bug: None
Change-Id: I4c6875c693350b95b8742a6a8e17157743db62cb
Reviewed-on: https://webrtc-review.googlesource.com/17400
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20502}
This reverts commit a7678667fc726979ca27d2429689d5735cca425d.
Reason for reland: Fix initializer list constructor.
Original change's description:
> Revert "Add helper functions for migrating to new video codec factories"
>
> This reverts commit 1c9623c70db42550d152c127e01434004087f743.
>
> Reason for revert: Breaks brace initialization:
>
> cricket::VideoDecoderParams params = { "deadbeef" };
>
> I suggest adding an initializer list constructor.
>
> Original change's description:
> > Add helper functions for migrating to new video codec factories
> >
> > This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> > convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> > webrtc::VideoEncoder and webrtc::VideoDecoder.
> >
> > The purpose is to make it as easy as possible for clients to migrate to the new
> > API and allow us to stop depending on the internal SW codecs as soon as possible.
> >
> > There still exists an ugly decoder adapter class in the video engine. The reason
> > is that we need to continue to pass in the |receive_stream_id| decoder params to
> > some legacy clients.
> >
> > Bug: webrtc:7925
> > Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> > Reviewed-on: https://webrtc-review.googlesource.com/15181
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20475}
>
> TBR=magjed@webrtc.org,andersc@webrtc.org
>
> Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/17160
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20486}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
Change-Id: Ic825d133b6e1c6e5aad811ba528751dd5ed85e67
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20501}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
Intended to enable unit testing of the pacer with a mock PacketQueue.
Bug: webrtc:8422
Change-Id: I142386b2d91ad0d5ba8f3f9d876e67972c490de4
Reviewed-on: https://webrtc-review.googlesource.com/17300
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20498}
Issue was found by the Audio Processing fuzzer.
Bug: chromium:778939, chromium:778921, chromium:778919
Change-Id: If613cf4c533f546d118f10a6358cecd329958177
Reviewed-on: https://webrtc-review.googlesource.com/16161
Commit-Queue: Alex Loiko <aleloi@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20494}
This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and
checks the output PSNR.
- 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: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
This reverts commit 1c9623c70db42550d152c127e01434004087f743.
Reason for revert: Breaks brace initialization:
cricket::VideoDecoderParams params = { "deadbeef" };
I suggest adding an initializer list constructor.
Original change's description:
> Add helper functions for migrating to new video codec factories
>
> This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> webrtc::VideoEncoder and webrtc::VideoDecoder.
>
> The purpose is to make it as easy as possible for clients to migrate to the new
> API and allow us to stop depending on the internal SW codecs as soon as possible.
>
> There still exists an ugly decoder adapter class in the video engine. The reason
> is that we need to continue to pass in the |receive_stream_id| decoder params to
> some legacy clients.
>
> Bug: webrtc:7925
> Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> Reviewed-on: https://webrtc-review.googlesource.com/15181
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20475}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20486}
The method prototype is being changed to returning a const char*,
we don't rely on it in webrtc and the point of the DCHECK that
previously referenced it, was to avoid usage of std::string for
histogram names.
Bug: webrtc:8472
Change-Id: I69b588d4a8f339911a051fd232d63ea5bb1f9a45
Reviewed-on: https://webrtc-review.googlesource.com/16940
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20484}
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
