Otherwise we will use 48kHz as default, the MSG will resample as needed.
It would be possible to allow all frequencies in the AudioConduit as the webrtc backend supports them all, however it would require more changes and likely heap allocation that we're trying to limit in this part of the code.
MozReview-Commit-ID: B3x5t1FSaQ8
--HG--
extra : rebase_source : 77f83a876ed9b5ded45419245655709aee2573df
Also remove the unmatched candidates in the candidate pair table since
they are now included in the "all raw candidates" table.
MozReview-Commit-ID: 4ZvhWfmjGJh
--HG--
extra : rebase_source : bf999db83cd49dd454434d2b157023da41b0dbcd
Adding trickle field that will allow us to flag trickled candidates
on about:webrtc.
Also added label field to NrIceCandidate to facilitate showing the
raw candidate info on about:webrtc.
MozReview-Commit-ID: HuP3IxYOOBJ
--HG--
extra : rebase_source : 975cb5b29b2aef233f856bfbdc8c325535d24272
This is a workaround for the fact that our code is wrong.
FakeAudioDeviceModule :::TimeUntilNextProcess() returns 0, so we're busy
looping.
I'm switching that to 100, which is arbitrary, but makes the problem go away. I
want to fix that soon, because this is a terrible workaround.
FakeAudioDeviceModule is _not_ made to be used in production.
MozReview-Commit-ID: FoGs6GFsRRN
--HG--
extra : rebase_source : d62f7c2d5b98c4814c06be13aca49bd478d5b381
This allows to re-use the SharedThreadPool across calls, preventing the need to create a new thread on each call.
MozReview-Commit-ID: CbP6OTYKhHL
--HG--
extra : rebase_source : 969f2c74f00614d6265fe0e25abfb36c9648d564
There's no need for a custom class to perform this task.
MozReview-Commit-ID: JxpDQVM97fl
--HG--
extra : rebase_source : 8387efa0ed3add9e4a42daed98e97372d9cabee5
We keep the synchronous version that will be used in bug 1424653
MozReview-Commit-ID: JTGaRYm20ca
--HG--
extra : rebase_source : aa694b7ac4f03322edbdefd64cefd198c0909cec
It allows to more easily distinguish between methods from upstream webrtc.org and our code.
MozReview-Commit-ID: ILQhEAYbSmc
--HG--
extra : rebase_source : 934505afddcca9253b00c4094776c95a087058fb
Now that the graph rate match the one out of NetEQ, we can remove an unecessary conversion.
Additionally, move a member from the base case to the only one where it's used.
MozReview-Commit-ID: II5mdcl0vhK
--HG--
extra : rebase_source : 1d9edfc2803c3fadde7505b4d84293640e4311e0
Also, pass arguments are const reference.
We also rename class members as per coding style.
MozReview-Commit-ID: 9IkV8wCMpz7
--HG--
extra : rebase_source : 6dc8285342742bf19dd2d03f66dd0668fc32bbcc
It removes the need to explicitly shutdown the taskqueue and wait on the taskqueue to have run all dispatched task.
We do want to enforce that no listeners are being called once the VideoFrameConverter's owner has been destroyed as it could potentially lead to a UAF.
For now, access is okay as all operations are performed on the MSG's thread.
However, this will change in follow up patches.
The SourceMediaStream keeps a raw pointer to the MSG, and check if it's value isn't null to determine if the MSG has been shutdown or not, however SourceMediaStream::mGraph isn't thread safe as its access isn't protected by a mutex/monitor.
MozReview-Commit-ID: 1QsJAzPuE6L
--HG--
extra : rebase_source : 35f34450e62ff1f445ad6ccd06c6f6cbd3c6ed54
It's bound to be done automatically, makes it easier to modify later.
MozReview-Commit-ID: IQ5TBtS8Z3v
--HG--
extra : rebase_source : 0b1a326ada5761cfa50c33acbff9b61cf4e59c8c
The MSG provides the reverse stream, and feed it directly to the APM.
MozReview-Commit-ID: A6DO407CJkp
--HG--
extra : rebase_source : df4ad965c171eab5a72a8d09e0305b1e79325a03
extra : source : e92ff1339db1ca5affa56ccdbec1c8b3836bcd95
This forces us to do a copy. It's not the end of the world but could be avoided.
The number of channels received is now explicit (via
`AudioFrame::num_channels_`), instead of being guessed based on the number of
samples (considering we're always dealing with 10ms of audio, and we know the
rate).
It's still coupled a bit with audio devices, but we cheat, and use a "fake audio
device", which isn't going to touch actual OS APIs.
MozReview-Commit-ID: 1Tfajkv1HQR
--HG--
extra : rebase_source : c0c8c240621b076bb3b056689f45289212498903
extra : source : 9e92591ba6dcb18364da98756c645c91bfe81517
We used to fix the rate, arbitrarily, to 32kHz. Because the graph is almost
never running at 32kHz (more like 44.1kHz or 48kHz), and the codec would often
not be at 32kHz, this meant multiple resampling:
- Once here, in MediaPipeline, to bring to 32kHz
- Once when getting inserted in the MSG (so that the audio was brought back to
MSG rate)
- Maybe once in cubeb (depending on the platform)
This always removes the second resampling: the track is now at the correct rate,
as far as the MSG is concerned.
Additionally, if the MSG is running at 48kHz, more resampling are saved, because
it's one of the native webrtc.org rates.
MozReview-Commit-ID: DBWcwuWxUpu
--HG--
extra : rebase_source : 2b961a8bd91d952ccbe9df5a6ab7649321f282a6
extra : source : a3d9aa2649b95329d0cf686d79aa5179e9f3506d
Csoregi Natalia
JW Wang
Jean-Yves Avenard
Heiher
Margareta Eliza Balazs
Matthew Gregan
Nils Ohlmeier [:drno]
Byron Campen [:bwc]
Michael Froman
Alex Chronopoulos
Paul Adenot
Sebastian Hengst
Sylvestre Ledru
Mike Hommey
Bogdan Tara