Only three items in the (rather large) header were actually used after
InsertPacket: payloadType, timestamp and sequenceNumber. They are now
put directly into Packet. This saves 129 bytes per Packet that no
longer need to be allocated and deallocated.
This also works towards decoupling NetEq from RTP. As part of that,
I've moved the NACK code earlier in InsertPacketInternal, together
with other things that directly reference the RTPHeader.
BUG=webrtc:6549
Review-Url: https://codereview.webrtc.org/2411183003
Cr-Commit-Position: refs/heads/master@{#14658}
for consistency with other rtcp packet classes.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2361853002
Cr-Commit-Position: refs/heads/master@{#14648}
and thus IP_PACKET_SIZE constant:
Build() use BlockLength() instead of constant IP_PACKET_SIZE for packet
capacity, adding extra checks about packet generation in tests.
Build(callback) removed as unused.
definitions reordered to follow style.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2270753002
Cr-Commit-Position: refs/heads/master@{#14647}
When the FlexfecReceiver recovers media packets, it inserts these into
internal::Call, which then distributes them to the appropriate
VideoReceiveStream/RtpStreamReceiver.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2390823009
Cr-Commit-Position: refs/heads/master@{#14642}
AEC and AECM when these become full to also work when not
in debug mode.
BUG=webrtc:6530
Review-Url: https://codereview.webrtc.org/2419023002
Cr-Commit-Position: refs/heads/master@{#14637}
To ensure this change won't break Chromium, this is the first change, to add a
new CaptureFrame() function, and let Capture(DesktopRegion) and CaptureFrame()
call each other. So both a legacy consumer or a legacy implementation won't be
broken.
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=6513
Review-Url: https://codereview.webrtc.org/2409833002
Cr-Commit-Position: refs/heads/master@{#14635}
With this change, the calculations inside AverageIAT are changed to be
in double-precision floating point instead of in fixed point. Also,
the method's name is changed to EstimatedClockDriftPpm to better
reflect what it returns.
A few unit tests had to be updated because of minor numerical
differences.
Also removing the UBSan suppression related to this issue.
BUG=webrtc:5889
Review-Url: https://codereview.webrtc.org/2408653002
Cr-Commit-Position: refs/heads/master@{#14628}
This was an ill tested special case which turned out to be more problem
than benefit. The special case was only triggered when the decoder frame
size was smaller than 10 ms, which is more or less unsupported by NetEq.
Also fixed a bug in a test, a bug which was exposed by the code change.
BUG=chromium:654983
Review-Url: https://codereview.webrtc.org/2412883002
Cr-Commit-Position: refs/heads/master@{#14627}
Reason for revert:
There is a risk of ending up in a bad state due to race conditions with this patch. Tests in downstream clients have shown that it can
happen that an output stream is opened up in MUSIC mode when it should not.
Reverting since the new functionality added here is not worth the
risk of breaking existing clients.
Original issue's description:
> Android audio playout now supports non-call media streams.
>
> 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
>
> Committed: https://crrev.com/872f614111f436d15e29516ce19c3b63d25b8639
> Cr-Commit-Position: refs/heads/master@{#14613}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2420583002
Cr-Commit-Position: refs/heads/master@{#14626}
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}
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}
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}
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}
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}
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}
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}
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}
Updating GN files, include paths, and include guards
BUG=None
NOTRY=True
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2387113005
Cr-Commit-Position: refs/heads/master@{#14542}
