Introduce FrameGeneratorInterface to make FrameGenerator API available
for downstream projects.
Bug: webrtc:10138
Change-Id: I4216775e4b8b54c3f1c72d67ffbda31eb082fd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161234
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30009}
This makes the class easier to use at a minor cost of making it slightly
more magic.
Bug: webrtc:9883
Change-Id: If807cfbf046615333c3bcd3b58a001813102a9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30008}
This change finally wires up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.
Bug: chromium:1013590
Change-Id: I76f8226752294aee8fe137d1a78ee66548900cc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161095
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30003}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
When building certain Chromecast build flavors in component build mode,
there are some link errors due to symbols not being exported. This CL
fixes those issues.
TBR: kwiberg@webrtc.org
Bug: None
Change-Id: I408f0a84b8ac4610cc6b5aa6ff58248ea82c9c66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161148
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29992}
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.
Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames
can flow to sinks installed in VideoTrackSourceInterface.
Bug: chromium:1013590
Change-Id: I0779932c251a2159880a39b2d42d5ce439cc88e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161090
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29988}
It should fix compilation errors that happen on some iOS bots saying
"definition of implicit copy assignment operator for 'Foo'
is deprecated because it has a user-declared copy constructor"
Bug: webrtc:11162
Change-Id: Ife3d1a800ed6a4cd08bdfd156cd0e320504ee8dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161221
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29984}
This CL adds code for doing signal-dependent downmixing
before the delay estimation in the multichannel case.
As part of the CL, the unittests of the render delay
controller are corrected. However, as that caused some of
them to fail, the CL (for now) as well disables the failing
test.
Bug: webrtc:11153,chromium:1029740, webrtc:11161
Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29980}
This CL decouples NetEqFactory and AudioDecoderFactory.
AudioDecoderFactory is used in more places than just inside of NetEq, so
decoupling these makes sense.
Bug: webrtc:11005
Change-Id: I78dd856e4248e398e69a65816b062ef30555b055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29961}
Since this macro can be considered public, it makes sense to prefix it
with WEBRTC_ (also to avoid potential conflicts with client code).
This CL also removes some definitions of this macro in order to define
it only where it is strictly needed (it is only used in a .cc file).
Bug: webrtc:11142
Change-Id: Idce7389301e71d8434e238b3cf4ceaa9cf97cd87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161008
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29957}
This CL changes the downmixing of the input to the delay estimation
for surround/stereo signals to be off by default.
A kill-switch is also added for enforcing the downmix to be on.
Bug: webrtc:10913
Change-Id: I1030fef593ba56416deeb13b80d2f3812bffb9ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161012
Commit-Queue: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29951}
This patch introduces 3 new functions on StunMessages
- Clone, copy a message
- IsStunMethod, verifies that a buffer is a StunMessage
w/o requring a fingerprint
- EqualAttributes, compare attributes in two stun messages
(with filter)
This methods will be used to implement GOOG_PING
BUG=webrtc:11100
Change-Id: I284726c74aa0437be0bb9fbcf943c7d64a18acec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160281
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29950}
There are edge cases where the caching of encoder info will cause
issues. For instance if a sub-encoder fails en Encode call and falls
back to some other implementation, or if the fps targets shift due to
SetRates() triggering new layers to be enabled.
This CL forces a complete rebuild on every call to GetEncoderInfo().
It also adds new logging of when the info changes, as debugging issues
can be very time consuming if we can't tell that happened.
Bug: webrtc:11000
Change-Id: I7ec7962a589ccba0e188e60a11f851c9de874fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29938}
Use that conversion instead of duplicating it in call/
Bug: webrtc:11042
Change-Id: I035b161d429ec339dd2ad9e9ed3ede5045fb6199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160881
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29936}
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.
Reason for revert: crbug.com/1028937
Original change's description:
> Add support for RtpEncodingParameters::max_framerate
>
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}
TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
Many WebRTC users need only Opus, and no other audio codecs. This
makes it convenient for them to do the right thing.
To prove that the new factories work, use them in
PeerConnectionEndToEndTest.
Bug: webrtc:11130
Change-Id: I2c2450ba0fb33ef3b50da8f6cd325cad6b1e59a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160648
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29921}
The NetEqFactory is currently expected to wrap the AudioDecoderFactory,
but this turns out not to be a good idea. Instead, it makes more sense
to pass the AudioDecoderFactory through the CreateNetEq method.
Bug: webrtc:11005
Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29918}
This change implements the methods in VideoTrackSourceInterface
that are related to encoded output.
Bug: chromium:1013590
Change-Id: Id9ddbc00a7098e9b44cee1517c69002865a5fb33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159926
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29912}
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.
Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
Implements an alternative to the dominant nearend detector.
Bug: b/130016532
Change-Id: If4867d58aad036ccf4e456ef81689b8db0284f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159865
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29906}
Creates an abstraction for an "alarm clock" which can schedule
time-controller callbacks and exposes a time controller driven by
an external alarm.
Bug: webrtc:9719
Change-Id: I08c2aa9dba25603043bfba48f55c925716a55bae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158969
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29879}
All downstream users have been moved to the new one.
Bug: webrtc:11091
Change-Id: Ia18d0df94a7b95b1a58b4a53cfb195c61ef59ffd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160201
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29873}
This patch adds
- Attribute: STUN_ATTR_GOOG_MESSAGE_INTEGRITY_32
which is a ordinary message integrity but truncated to 32-bit
- Method: GOOG_PING,
which will be used for webrtc:11100
Both the attribute and the method has been registered at iana,
https://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml#stun-parameters-4
BUG=webrtc:11100
Change-Id: Iddd5614473fd6f18fbbe76e72d047c617df7123f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160180
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29864}
This reverts commit e6eded31e642b3b986fef478315603b5f398c227.
Reason for revert: A better method for communicating encoded frames in VideoTrackSourceInterface surfaced.
Original change's description:
> VideoFrame: Store a reference to an encoded frame
>
> Enable webrtc::VideoFrame to store a reference to an encoded frame.
>
> Bug: chromium:1013590
> Change-Id: Id5a06f1c7249f104dfd328f08677cf8001958f0d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158788
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29809}
TBR=ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,handellm@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1013590
Change-Id: I46384b7997e7b1cd3a2a2042cf17890fc977cca3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160204
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29863}
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.
Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
This change adds a new type of sink for consuming encoded data from
a video source.
Bug: chromium:1013590
Change-Id: Ia7c4e372190c3d6bc007a0d4deb05c2d1bce58d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159927
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29856}
This part of the effort to implement A/V sync metric.
Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
Add tests for different UpdateRect methods as they are no longer trivial
This change will enable providing useful update rects after scaling
is done.
Bug: webrtc:11058
Change-Id: I2311dbbbb5eca5cfaf845306674e6890050f80c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159820
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29835}
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).
I checked what our downstream users are actually using, and it's
cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage
I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.
There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.
Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.
Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
This is a reland of 53e157d25ce78ba6cd8625b0b655b46f8e1b0a91
The issue has been fixed in
https://chromium-review.googlesource.com/c/chromium/src/+/1917204.
Original change's description:
> Force Chromium deps on the WebRTC component.
>
> This CL adds a visibility check to the rtc_* GN templates in order
> to force Chromium to depend only on publicly visible targets from
> //third_party/webrtc_overrides and not from //third_party/webrtc.
>
> This is required in order to ensure that the Chromium's component
> builds continues to work correctly without introducing direct
> dependency paths on WebRTC that would statically link it in multiple
> shared libraries.
>
> Bug: webrtc:9419
> Change-Id: Ib89f4fc571512f99678ee4f61696b316374346d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154344
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@chromium.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29806}
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I7123d1b44ddbc23b11d9fa25aa39aa420359e33d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159922
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29816}
