The main reason for doing this is to allow refcounted objects to accept rvalue references in ctor and be able to std::move ctor rvalue arguments.
Also, refcounted.h is now generated using pump.py instead of manually creating each ctor version.
BUG= none
Review-Url: https://codereview.webrtc.org/2425683003
Cr-Commit-Position: refs/heads/master@{#14687}
Reason for revert:
Breaks internal project
Original issue's description:
> Move current bitstream parser to more appropriate directory.
>
> This CL groups together the code that has to do with parsing H264 bitstreams.
> This code logically belongs together, and having it in the same directory not
> only simplifies things from a project structure perspective, but also makes it
> easier to refactor out common parts incrementally.
> An added benefit is that this simplifies modular compilation, where for example
> one would like a build of WebRTC without the H264 codec-specific parts.
>
> BUG=webrtc:6338
>
> Committed: https://crrev.com/cc6817e9ce4a5ffc73efb660cf0368afbc7d9a4f
> Cr-Commit-Position: refs/heads/master@{#14684}
TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6338
Review-Url: https://codereview.webrtc.org/2430353004
Cr-Commit-Position: refs/heads/master@{#14685}
This CL groups together the code that has to do with parsing H264 bitstreams.
This code logically belongs together, and having it in the same directory not
only simplifies things from a project structure perspective, but also makes it
easier to refactor out common parts incrementally.
An added benefit is that this simplifies modular compilation, where for example
one would like a build of WebRTC without the H264 codec-specific parts.
BUG=webrtc:6338
Review-Url: https://codereview.webrtc.org/2370853005
Cr-Commit-Position: refs/heads/master@{#14684}
ScreenCapturer tests may fail on trybot, so this change is to fix the issue.
Changes include,
1. Sometimes, a capturer may capture part of the change, i.e. usually the draw
actions are not atomic. So the updated_region may be inaccurate. So I have added
a MayDrawIncompleteShapes() function in ScreenDrawer. If it returns false, the
updated_region check will be ignored.
2. Several test cases may run concurrently, which makes one ScreenDrawer won't
really work. Its window may be covered by another ScreenDrawer. So I have added
a system wide lock to ensure only one ScreenDrawer is working at a certain time.
3. On unity (Linux), the top several pixels of a window may be covered by a
shadow effect if the window is not focused. So I have added a BringToFront()
function, and call it in WaitForPendingDraws().
4. On Windows, the drawn shapes are 'temporary drawing', which will be erased
once the window is covered by another one. So I repeat DrawRectangle() function
call in the test case.
TODO(zijiehe): The DISABLED_ prefixes will be added back after the code review.
And I will move these test cases into modules_test in a coming change.
BUG=647067
Review-Url: https://codereview.webrtc.org/2337073007
Cr-Commit-Position: refs/heads/master@{#14674}
The main goal of this CL is to remove old buffer handling using static arrays
and switch to the improved rtc::Buffer class instead.
By doing so, we can remove some members (since Buffer maintains them instead) and
do some additional cleanup.
This CL also fixes some minor style issues and improves the locking mechanism.
Finally, AudioDeviceBuffer::SetRecordingChannel() is deprecated since it has never been
used and is not included in any test.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2333273002
Cr-Commit-Position: refs/heads/master@{#14661}
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}
