Make sure that the appropriate run loop source gets added/removed. More clean up
to remove unnecessary functions and suppress deprecated declaration warnings.
BUG=webrtc:6029
Review-Url: https://codereview.webrtc.org/2417603002
Cr-Commit-Position: refs/heads/master@{#14615}
This gets rid of a bit of codec-specific code in VoE.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
The solution is somewhat experimental.
NOTRY=TRUE
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2411263003
Cr-Commit-Position: refs/heads/master@{#14613}
Changed mixability status into AddSource/RemoveSource. Added 'ssrc()'
method to the MixerSource interface. Removed unnecessary member 'num_audio_sources_' and made the mixer be refcounted.
BUG=webrtc:6346
NOTRY=True
Review-Url: https://codereview.webrtc.org/2408683002
Cr-Commit-Position: refs/heads/master@{#14612}
to the functionality in the audio processing module.
Therefore, it should be a pure interface.
This CL ensures that is the case.
BUG=webrtc:6515
Review-Url: https://codereview.webrtc.org/2406193002
Cr-Commit-Position: refs/heads/master@{#14608}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Prune connections based on network name.
> Previously we prune connections on the same network pointer.
> So if an IPv6 and an IPv4 network are on the same network interface, IPv4 connection won't be pruned even if an IPv6 connection with higher priority becomes writable.
>
> With this change, as long as one connection becomes writable, all connections having lower priority with the same network name will be pruned.
>
> Also simplify the implementation.
>
> BUG=webrtc:6512
>
> Committed: https://crrev.com/aae2784c1fab9d1510393dec15d76caa574e2da8
> Cr-Commit-Position: refs/heads/master@{#14593}
TBR=skvlad@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6512
Review-Url: https://codereview.webrtc.org/2412433003
Cr-Commit-Position: refs/heads/master@{#14601}
MixerAudioSource is moved to AudioMixerImpl::Source. Structures and methods of the MixerAudioSource interface have been renamed. The RemixFrame method has added checks and is moved to audio_frame_manipulator.h
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396803004
Cr-Commit-Position: refs/heads/master@{#14600}
After calling Start(), doing a Stop() then Start() sequence should bring
the stream back to the original state.
BUG=webrtc:6501
Review-Url: https://codereview.webrtc.org/2407163002
Cr-Commit-Position: refs/heads/master@{#14597}
Compromise solution where WebRtcAudioUtils.setWebRtcBasedAutomaticGainControl() is marked
as deprecated and where as many APIs as possible that touches the HW AGC are removed. Some basic architecture is saved to ensure that we can restore usage of
the HW AGC if ever required for future devices.
The AppRTCMobile demo does still contain an AGC check box but it is now grayed out.
BUG=b/30387905
Review-Url: https://codereview.webrtc.org/2402883003
Cr-Commit-Position: refs/heads/master@{#14596}
This class is split in interface/implementation classes, since it
will be referenced from the Call level. Its purpose is to interface
the erasure code decoder with a new class FlexfecReceiveStream
(for received packets), as well as with the main RTP pipeline (for
recovered packets).
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2392663006
Cr-Commit-Position: refs/heads/master@{#14594}
Previously we prune connections on the same network pointer.
So if an IPv6 and an IPv4 network are on the same network interface, IPv4 connection won't be pruned even if an IPv6 connection with higher priority becomes writable.
With this change, as long as one connection becomes writable, all connections having lower priority with the same network name will be pruned.
Also simplify the implementation.
BUG=webrtc:6512
Review-Url: https://codereview.webrtc.org/2395243005
Cr-Commit-Position: refs/heads/master@{#14593}
It would be enough to say we're removing EnableSrtpDebugging because
it's never called, but the story is a bit more interesting.
libsrtp's debugging facilities are gated behind the reasonably-named
ENABLE_DEBUGGING macro:
b17c065a8a/srtp/crypto/include/err.h (186)
This code was imported to WebRTC from libjingle, but neither WebRTC or
Chromium ever set ENABLE_DEBUGGING. Even if someone had ever called
EnableSrtpDebugging, it wouldn't have done anything.
BUG=0
Review-Url: https://codereview.webrtc.org/2409513002
Cr-Commit-Position: refs/heads/master@{#14592}
This was missed in the first pass because this code only compiles in
Chromium.
BUG=webrtc:6376
Review-Url: https://codereview.webrtc.org/2407743002
Cr-Commit-Position: refs/heads/master@{#14591}
CGRegisterScreenRefreshCallback (and similar) have been replaced by
CGDisplayStream.
Most of the structure is pretty comparable. The main difference is that a
CGDisplayStream needs to be destroyed asynchronously, potentially after
ScreenCapturerMac has been destroyed. This CL creates a self-owned
DisplayStreamManager which will destroy itself once all streams have been
destroyed.
BUG=webrtc:6029
Review-Url: https://codereview.webrtc.org/2391743004
Cr-Commit-Position: refs/heads/master@{#14590}
Gestalt has been deprecated since macOS 10.8, and it's always been overkill for
finding the macOS version anyways. uname works fine.
BUG=webrtc:6027
Review-Url: https://codereview.webrtc.org/2391633004
Cr-Commit-Position: refs/heads/master@{#14589}
Make sure that WEBRTC_VOICE_ENGINE_AGC, WEBRTC_VOICE_ENGINE_ECHO, and
WEBRTC_VOICE_ENGINE_NR are always defined.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2401393002
Cr-Commit-Position: refs/heads/master@{#14587}
- Rename the data codec payload types to end with "PlType" instead of "Id", for consistency.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2397413002
Cr-Commit-Position: refs/heads/master@{#14581}
The rotation is currently incorrect for the back camera in landscape
mode. The reason is that device rotation needs to be reversed for the
back camera compared to the front camera. The camera sensor can also be
mounted with a specific orientation. So when front camera rotation goes
through: 0->90->180->270, back camera rotation goes in reverse:
180->90->0->270.
BUG=b/31984246,b/30651939
Review-Url: https://codereview.webrtc.org/2401033002
Cr-Commit-Position: refs/heads/master@{#14580}
receive a signal level to use initially, instead of the
default initial signal level.
The initial form of the CL
(https://codereview.webrtc.org/2254973003/) was reverted
due to down-stream dependencies. These have been resolved,
but the CL needed to be revised according to the new scheme
for passing parameters to the audio processing module.
Therefore, please review this CL as if it is new.
TBR=aleloi@webrtc.org
BUG=webrtc:6386
Review-Url: https://codereview.webrtc.org/2337083002
Cr-Commit-Position: refs/heads/master@{#14579}
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.
This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).
BUG=webrtc:6393
Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
The decoder implementation may have its own thread for asynchrouns
callbacks. We need to stop it (by releasing the decoder) when stopping
the decoder thread, othweise it may call incoming_video_stream_ after
it has been destroyed.
BUG=webrtc:6501
Review-Url: https://codereview.webrtc.org/2402663003
Cr-Commit-Position: refs/heads/master@{#14577}
all methods but a few constructors. And similarly for the
subclass cricket::WebRtcVideoFrame.
TBR=tkchin@webrtc.org # Added an include line
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2315663002
Cr-Commit-Position: refs/heads/master@{#14576}
The values in question are supposed to be able to be negative.
BUG=chromium:653448
Review-Url: https://codereview.webrtc.org/2387333005
Cr-Commit-Position: refs/heads/master@{#14573}
The purpose is to prepare for a TextureViewRenderer that will share the
EGL rendering code.
Two functional changes are also included:
* The implementation of SurfaceHolder.Callback.surfaceDestroyed will now
block until the EGL surface is released. This is done in order to
comply with the documentation that says: "If you have a rendering
thread that directly accesses the surface, you must ensure that thread
is no longer touching the Surface before returning from this function."
* We will no longer try to hide render glitches during layout changes.
This was a lost cause anyway.
BUG=webrtc:6407
Review-Url: https://codereview.webrtc.org/2399463006
Cr-Commit-Position: refs/heads/master@{#14570}
Reason for revert:
breaks chromium FYI
Original issue's description:
> Made MixerAudioSource a pure interface.
>
> This required quite a few small changes in the mixing algorithm
> structure, the mixer interface and the mixer unit tests.
>
> BUG=webrtc:6346
>
> Committed: https://crrev.com/2ae5fdff86b784545cbd724de54bb5ffedde1adf
> Cr-Commit-Position: refs/heads/master@{#14567}
TBR=ivoc@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2394253003
Cr-Commit-Position: refs/heads/master@{#14568}
This required quite a few small changes in the mixing algorithm
structure, the mixer interface and the mixer unit tests.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396483002
Cr-Commit-Position: refs/heads/master@{#14567}
