This CL finalizes the support for allowing an external
audio processing module to be used in a peerconnection.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2965703002
Cr-Commit-Position: refs/heads/master@{#18864}
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
This was previously reverted, because external projects were using the
internal webrtc::AudioEncoderOpus class and broke when it was renamed.
This re-land avoids renaming it immediately, to give those projects
time to adapt. It also has to revert some of the changes I had made to the
Config struct, since that was also used by the same external projects.
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2948483002
Cr-Commit-Position: refs/heads/master@{#18852}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
magjed has written most of the code in this folder.
NOTRY=TRUE
Bug: None
Change-Id: I786261d4407f38de612f5fae12b9abde4594bac2
Reviewed-on: https://chromium-review.googlesource.com/550095
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18829}
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
Reason for revert:
Breaking google3 projects
Original issue's description:
> Opus implementation of the AudioEncoderFactoryTemplate API
>
> Now the templated AudioEncoderFactory can create Opus encoders!
>
> BUG=webrtc:7831
>
> Review-Url: https://codereview.webrtc.org/2930243003
> Cr-Commit-Position: refs/heads/master@{#18645}
> Committed: fe1aa82c63TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2947563002
Cr-Commit-Position: refs/heads/master@{#18649}
Now the templated AudioEncoderFactory can create Opus encoders!
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2930243003
Cr-Commit-Position: refs/heads/master@{#18645}
Now the templated AudioEncoderFactory can create G722 encoders!
BUG=webrtc:7833
Review-Url: https://codereview.webrtc.org/2934833002
Cr-Commit-Position: refs/heads/master@{#18644}
Now the templated AudioDecoderFactory can create G722 decoders!
BUG=webrtc:7839
Review-Url: https://codereview.webrtc.org/2940833002
Cr-Commit-Position: refs/heads/master@{#18643}
No real encoder implements the correct API yet, so we're just testing
dummies.
BUG=webrtc:7823
Review-Url: https://codereview.webrtc.org/2935643002
Cr-Commit-Position: refs/heads/master@{#18637}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
I keep having to re-write these whenever I'm debugging.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2936533003
Cr-Commit-Position: refs/heads/master@{#18586}
Reason for revert:
Broken downstream project.
Original issue's description:
> Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
>
> BUG=webrtc:7395
>
> Review-Url: https://codereview.webrtc.org/2888303005
> Cr-Commit-Position: refs/heads/master@{#18417}
> Committed: 9641c13327TBR=deadbeef@webrtc.org,stefan@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,holmer@google.com,zstein@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2914413002
Cr-Commit-Position: refs/heads/master@{#18420}
Previously, the base class PlanarYuvBuffer was used directly. Having
separate base classes will allow us to improve type safety in some
places.
BUG=webrtc:7632
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2914463002
Cr-Commit-Position: refs/heads/master@{#18317}
probing_interval as a name is used for the period that BWE attempt to increase its estimate. The name is confusing since it is not related to "probing" which is a special mechanism for estimating BWE.
BUG=None
Review-Url: https://codereview.webrtc.org/2888893002
Cr-Commit-Position: refs/heads/master@{#18203}
Add the include to the files where it is actually used instead.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2869863003
Cr-Commit-Position: refs/heads/master@{#18176}
Reason for revert:
Re-land the original CL because the reverting it didn't fix the problem.
Original issue's description:
> Revert of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2873803002/ )
>
> Reason for revert:
> Reverted because it possibly breaks the internal project.
>
> Original issue's description:
> > Make AudioSinkInterface::Data hold a const pointer to the audio buffer.
> >
> > This is in preparation for https://codereview.webrtc.org/2750783004/, where
> > requiring a non-const pointer for AudioSinkInterface would force an unmuting
> > and zeroing of every frame.
> >
> > BUG=webrtc:7343
> >
> > Review-Url: https://codereview.webrtc.org/2873803002
> > Cr-Commit-Position: refs/heads/master@{#18107}
> > Committed: 38605965bd
>
> TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7343
>
> Review-Url: https://codereview.webrtc.org/2877013002
> Cr-Commit-Position: refs/heads/master@{#18112}
> Committed: c904634823TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7343
Review-Url: https://codereview.webrtc.org/2880663003
Cr-Commit-Position: refs/heads/master@{#18113}
Reason for revert:
Reverted because it possibly breaks the internal project.
Original issue's description:
> Make AudioSinkInterface::Data hold a const pointer to the audio buffer.
>
> This is in preparation for https://codereview.webrtc.org/2750783004/, where
> requiring a non-const pointer for AudioSinkInterface would force an unmuting
> and zeroing of every frame.
>
> BUG=webrtc:7343
>
> Review-Url: https://codereview.webrtc.org/2873803002
> Cr-Commit-Position: refs/heads/master@{#18107}
> Committed: 38605965bdTBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7343
Review-Url: https://codereview.webrtc.org/2877013002
Cr-Commit-Position: refs/heads/master@{#18112}
VideoFrameBuffer is currently hard coded to be either I420 or Native.
This CL makes VideoFrameBuffer more generic by moving the I420 specific
functions into their own class, and adds an enum tag that represents the
format and storage type of the buffer. Each buffer type is then
represented as a subclass. See webrtc/api/video/video_frame_buffer.h for
more info.
This CL also adds support for representing I444 in VideoFrameBuffer
using the new interface. Possible future buffer type candidates are
RGB and NV12.
BUG=webrtc:7632
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2847383002
Cr-Commit-Position: refs/heads/master@{#18098}
I'm already a root OWNER, but people tend to pick reviewers from the
most specific OWNERS file, so I should probably be in these two.
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2869443002
Cr-Commit-Position: refs/heads/master@{#18038}
So that it's easier to find the right reviewers.
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2860093005
Cr-Commit-Position: refs/heads/master@{#18037}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
This will create another decoder instance, which isn't ideal, but
that's better than the current behavior where things don't work at all.
We still need to fix the root cause of the linked bug, which is that we
don't remember previous payload type mappings when generating an offer.
This CL also adds a check for what is a real error: when a payload type
that was mapped to one codec is changed to map to a different codec.
And lastly, this CL removes a DCHECK for an assumption that was not
valid: that subsequently applied codec lists will always be supersets of
previous lists.
BUG=webrtc:5847
Review-Url: https://codereview.webrtc.org/2831333002
Cr-Commit-Position: refs/heads/master@{#17897}
This target keeps track of .h the files under webrtc/modules/include/
that are not part of any target.
If a .h file is not part of a target the 'gn check' utility is not
able to spot if a target is missing a dependency because even if
it parses '#include' directives it is not able to find a target that
contains these headers.
BUG=webrtc:7513
NOTRY=True
Review-Url: https://codereview.webrtc.org/2838873002
Cr-Commit-Position: refs/heads/master@{#17880}
Make all rtc_source_test target that contains tests that
are included in a test executable only be visible to the
rtc_test target. Doing this exposed a couple of errors and
dependency problems that were resolved. Having this could
have prevented duplicated execution of tests like the case that
was recently fixed by deadbeef@ in
https://codereview.webrtc.org/2820263004
New targets:
* //webrtc/modules/rtp_rtcp:fec_test_helper
* //webrtc/modules/rtp_rtcp:mock_rtp_rtcp
* //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer
The mock files and targets should probably be moved into webrtc/test in
the future, but that's out of the scope of this CL.
BUG=webrtc:5716
NOTRY=True
Review-Url: https://codereview.webrtc.org/2828793003
Cr-Commit-Position: refs/heads/master@{#17863}
