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CGRegisterScreenRefreshCallback (and similar) have been replaced by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}
The RTCStatsCollector collects candidates from candidate pairs. Note
that there may be other candidates that are not paired with anything,
stats for these should also be produced before closing crbug.com/632723.
[1] https://w3c.github.io/webrtc-stats/#icecandidate-dict*
BUG=chromium:627816, chromium:632723
Review-Url: https://codereview.webrtc.org/2384143002
Cr-Commit-Position: refs/heads/master@{#14565}
but remove the #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE so that it always builds.
BUG=webrtc:6497
Review-Url: https://codereview.webrtc.org/2398123002
Cr-Commit-Position: refs/heads/master@{#14564}
Some runtime dependencies on MSVC were missing and had to be added to rtc_base_approved.
BUG=chromium:497757
NOTRY=True
Review-Url: https://codereview.webrtc.org/2389133002
Cr-Commit-Position: refs/heads/master@{#14562}
The default process interval (2000 ms) is used for the other counter classes.
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2388043003
Cr-Commit-Position: refs/heads/master@{#14560}
This makes this target not include the precompiled header, which
is different from GYP.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2402493002
Cr-Commit-Position: refs/heads/master@{#14559}
This CL deletes the old and not used video defines in
engine_configurations.h and pre-pends voice_ to indicate there are only
voice/audio defines left in the file.
BUG=none
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/2401673002 .
Cr-Commit-Position: refs/heads/master@{#14558}
The renaming is to reflect this class is only used for RTCP interaction
and not for other transports.
This Cl will be followed by multiple CLs moving all send-side RTP
functionality to a separate class, rtp module ownership away from
VideoSendStream and use TaskQueue instead of ProcessThread for RTP.
BUG=webrtc:6456
Review-Url: https://codereview.webrtc.org/2390463002
Cr-Commit-Position: refs/heads/master@{#14556}
code which is not thread-safe in the sense that the
rdft_init method can only be run in a single-threaded.
Currently, inside WebRTC multiple instances of the audio-
processing module are set up which means that the init
method may be run concurrently.
In order to avoid having to protect the init method with
a lock to ensure single-threaded behavior that, this CL
places the FFT functionality inside a class so that there
is no global component of the FFT functionality.
Note that:
1) The nonstandard header for the ooura_fft.cc was copied
from the aec_rdft.cc header, and augmented with a
description of the changes introduced in this CL.
2) The clang warnings for the ooura_fft_sse2.cc,
ooura_fft_neon.cc and ooura_fft_mips.cc were not
addressed as this code was kept as it was before this CL
3) Clang-format was run on all files apart from
ooura_fft_mips.cc (as that would change the format of
the inline assempbly code).
Adding bypass of presubmit to avoid code style and header errors caused by the fact that files with legacy code are being renamed.
NOPRESUBMIT=true
BUG=chromium:638583
Review-Url: https://codereview.webrtc.org/2348213002
Cr-Commit-Position: refs/heads/master@{#14554}
This should allow us to enable Intel HW VP8 encoder again.
BUG=webrtc:6232,b/30947951
Review-Url: https://codereview.webrtc.org/2263043003
Cr-Commit-Position: refs/heads/master@{#14552}
It eliminates the need for hardcoded enumerations of the supported presets.
Hopefully it enables us to support slightly bigger set of resolutions and make the code slightly cleaner.
BUG=webrtc:6355
Review-Url: https://codereview.webrtc.org/2349223002
Cr-Commit-Position: refs/heads/master@{#14547}
