Removes current_mic_level_, new_mic_level_ and clock_drift_, together
with APIs for accessing them.
Bug: webrtc:8598
Change-Id: I8e07396fcafd2a719e204730e2c7d26797bed762
Reviewed-on: https://webrtc-review.googlesource.com/39783
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21632}
These headers contain public defines for the module, so it makes
sense it's visible.
Bug: webrtc:7620
Change-Id: I4c2604dcfb3beb4bfa0803a15d0e1a89c374e4cf
Reviewed-on: https://webrtc-review.googlesource.com/39261
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21616}
Instead of keeping the umbrella header in sync manually and needing
ifdefs to make it include the correct headers depending on platform,
generate it based on the headers we include in the framework target.
Can also be used to only include internal software codec headers when
compiling with support for them.
Bug: webrtc:7925
Change-Id: I63f97af1efc8710cfd62d527fcb343fed05daae2
Reviewed-on: https://webrtc-review.googlesource.com/38702
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21613}
It should not be used anymore - see
https://webrtc-review.googlesource.com/39503
but it can be removed only after the DEPS roll.
Bug: webrtc:8745
Change-Id: I1ee29948e99be6cb4a700b67d37f72a0747a9fc5
Reviewed-on: https://webrtc-review.googlesource.com/39505
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21612}
API is not used anywhere for now. When depending projects implement it
or build with default implementation, it will be used.
Bug: webrtc:8287
Change-Id: I9e2aa922c2bb2b543793cd0561d797b02288ea6c
Reviewed-on: https://webrtc-review.googlesource.com/39042
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21611}
New OwnedPeerConnection takes ownership of the observer. This is done
to allow NativePeerConnectionFactory to return a capsulated object.
Bug: webrtc:8662
Change-Id: Ie876f7b9a1a17ebcfbe51537f712a32ab1a7cbfb
Reviewed-on: https://webrtc-review.googlesource.com/35300
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21610}
This version of stream stats will iterate over senders and
receivers and note which streams they think they know about,
rather than iterating over streams.
This means that streams mentioned in AddTrack() are also
included, and that only tracks actually attached are included
for those streams.
Bug: webrtc:8616
Change-Id: I4e704b1a47a152812f23a448cf1a6bc3af1ffafa
Reviewed-on: https://webrtc-review.googlesource.com/39262
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21609}
Similar to the change for RtpReceivers, this removes the BaseChannel
methods that would just proxy calls to the MediaChannel and instead
gives the MediaChannel directly to the RtpSenders to make the calls
directly.
Bug: webrtc:8587
Change-Id: Ibab98d75ff1641e902281ad9e31ffdad36caff35
Reviewed-on: https://webrtc-review.googlesource.com/38983
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21608}
We have done changes to the Audio Processing fuzzer here
https://webrtc-review.googlesource.com/c/src/+/36500/6.
We ran the new version of the fuzzer locally. The UBSAN
detector found these (minor) issues.
We have used the Godbolt compiler explorer to check that similar
changes produce identical compiled code.
Bug: webrtc:7820
Change-Id: I9cc3b81e4be7cf691f878c37010ce105bc2f3e38
Reviewed-on: https://webrtc-review.googlesource.com/39264
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21605}
This CL generalizes the hysteresis behavior on the AEC3 delay estimator
to be two-sided and easier to configure.
Bug: webrtc:8671
Change-Id: Ife21c1511416e32eb3618c81178deefe332ac1e8
Reviewed-on: https://webrtc-review.googlesource.com/39267
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21604}
This is a reland of https://webrtc-review.googlesource.com/34380
The main problem with that CL was that we used frame timestamps as basis
for frame dropping, but those might not be continuous or even populated
in some circumstances.
Additionally, I found that the bitrate was off since the encoder does
not not take the dropped frames into account, so if we drop every other
frame continiusoly, the bitrate sent will be around half of the target.
Patch set 1 is the original CL, subsequent patch sets cotains fixes.
Bug: webrtc:4172
Change-Id: I8ec8dddcebf4ce44f28dd9055cf9c46bbd68e4a6
Reviewed-on: https://webrtc-review.googlesource.com/39201
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21601}
This replaces most of the existing dependencies on the application
limited region(ALR) detector. This is to achieve a greater separation of
concerns and will make further refactoring regarding the ALR Detector
less invasive on other parts of the code base.
Bug: webrtc:8415
Change-Id: I92912254c6d02285cce6a88f6789f0ac94794c88
Reviewed-on: https://webrtc-review.googlesource.com/37560
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21598}
Now that voe::Channel is owned by Audio[Send|Receive]Stream, the legacy
VoiceEngine and the VoEBase interface is unused.
Also removes Atomic32, which was only used for ref counting VoiceEngine.
Bug: webrtc:4690
Change-Id: I73b8a083df544a8ab6383d57075a65ce955c592a
Reviewed-on: https://webrtc-review.googlesource.com/38723
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21595}
These new bots will check all that all WebRTC patches compile in
Chromium before landing. They will not run tests, but this should be
enough to catch the majority of the errors.
Expected cycle time is ~30 minutes, so they may slow down the CQ.
I will monitor this so the average cycle time isn't too adversely
affected.
Bug: chromium:666656
Change-Id: I09c0409ea614570b3c28bf4be3644dfffacb424c
Reviewed-on: https://webrtc-review.googlesource.com/39006
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21594}
GetSimulcastConfig had to be overloaded in order to not break downstream
client tests when the API was changed. Now that the downstream client
has been updated to use the new API, we can remove the overloaded
function.
Bug: webrtc:8630
Change-Id: I5d5d494e0579e60858d6efbb4715761394874b38
Reviewed-on: https://webrtc-review.googlesource.com/38882
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21590}
This is a reland of e357a4dd4e3b015f8281813f246de793589bd537
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org
Bug: webrtc:8673
Change-Id: I610302efc5393919569b77e3b59aa3384a9b88a5
Reviewed-on: https://webrtc-review.googlesource.com/38842
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21589}
This adds a generic interface for an echo detector, and makes it possible to inject one into the audio processing module.
Bug: webrtc:8732
Change-Id: I30d97aeb829307b2ae9c4dbeb9a3e15ab7ec0912
Reviewed-on: https://webrtc-review.googlesource.com/38900
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21588}
This is a reland of 046f78cae64fec756391e81206c5aa007274b791
Original change's description:
> Make freeNativePeerConnectionObserver generic.
>
> Previously, it was only possible to free PeerConnectionObserverJni
> objects using this method. Now it is generic and can free any
> PeerConnectionObserver.
>
> Bug: webrtc:8662
> Change-Id: I619ca5ed88a0c2553fa6d19ce41e510947d5bd44
> Reviewed-on: https://webrtc-review.googlesource.com/35222
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21411}
Bug: webrtc:8662
Change-Id: Iba64d613f7b434260a0d7b762ca67d49b295a84f
Reviewed-on: https://webrtc-review.googlesource.com/38901
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21587}
This reverts commit e357a4dd4e3b015f8281813f246de793589bd537.
Reason for revert: Looks like it's breaking some downstream projects.
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
Change-Id: I621c10236c02be01d82f4660168f0323b85e24af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8673
Reviewed-on: https://webrtc-review.googlesource.com/38681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21586}
