This is a trivial change, I just cutted and pasted part of the code in
screen_capturer_unittest.cc to screen_capturer_integration_test.cc, removed
DISABLED_ prefixes, and updated build file.
BUG=webrtc:6366
Review-Url: https://codereview.webrtc.org/2444583002
Cr-Commit-Position: refs/heads/master@{#14806}
This helps a lot to avoid reducing the bitrate too quickly when there's a short period of very few packets delivered, followed by the rate resuming at the regular rate. It specifically avoids the BWE going down to super low values as a response delay spikes.
BUG=webrtc:6566
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2422063002 .
Cr-Commit-Position: refs/heads/master@{#14802}
Enabled the plugin and cleaned up all issues it found, mainly virtual
destructors not being marked as override.
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2436503004
Cr-Commit-Position: refs/heads/master@{#14793}
- Update MockAudioProcessing to current APM interface.
- Replace calls to VoEAudioProcessing::Start/StopAecDump with direct calls on APM.
- Add AudioProcessing* in WVoE, get it from VoE, so we can call directly on APM.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2446143002
Cr-Commit-Position: refs/heads/master@{#14786}
Include CVO in key frame.
Include CVO in delta frame when rotation changes.
Include CVO when it is non zero to support current receiver implementation.
BUG=webrtc:6600
Review-Url: https://codereview.webrtc.org/2452583002
Cr-Commit-Position: refs/heads/master@{#14784}
Reason for revert:
Internal project has been fixed
Original issue's description:
> Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ )
>
> 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
>
> Committed: https://crrev.com/f04f14e772f803de39f8a6128e5157127cd35103
> Cr-Commit-Position: refs/heads/master@{#14685}
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/2434043002
Cr-Commit-Position: refs/heads/master@{#14783}
This refactoring allows runtime checks that functions that access
codec specific information are using the correct union member.
The API also allows replacing the union with another implementation
without changes at calling sites.
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2001533003
Cr-Commit-Position: refs/heads/master@{#14775}
Rename kFullSendSideEstimator -> kSendSideEstimator and add new class SendSideBweSender (controlled by kSendSideEstimator) that actually uses the send side BWE.
Move the mock to logging/rtc_event_log/mock.
Allow congestion_controller, remote_bitrate_estimator and audio to depend on loggging/rtc_event_log
BUG=webrtc:6526
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2431093003
Cr-Commit-Position: refs/heads/master@{#14772}
Several subcomponents inside APM copy render audio from
the render side to the capture side using the same
pattern. Currently this is done independently for the
submodules.
This CL moves the the AGC functionality for this into
APM.
BUG=webrtc:5298, webrtc:6540
Review-Url: https://codereview.webrtc.org/2444283002
Cr-Commit-Position: refs/heads/master@{#14770}
the render side to the capture side using the same
pattern. Currently this is done independently for the
submodules.
This CL moves the the AECM functionality for this into
APM.
BUG=webrtc:5298, webrtc:6540
Review-Url: https://codereview.webrtc.org/2444793005
Cr-Commit-Position: refs/heads/master@{#14768}
Remove functions to enumerate all extensions,
Remove concept of the inactive extension.
Decision if extension should be included into rtp header is done by rtp_sender
GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec.
Since extension might suddenly be included in the next packet (which still might belong to same fec group), it is safer to calculate all registered extension.
BUG=webrtc:5565, webrtc:1994
Review-Url: https://codereview.webrtc.org/2431253003
Cr-Commit-Position: refs/heads/master@{#14763}
The API has changed for the slice config of SSpatialLayerConfig as of
OpenH264 v1.6. Update H264EncoderImpl with an ifdef that uses the
correct API depending on what version of OpenH264 is being used.
BUG=webrtc:6583
Review-Url: https://codereview.webrtc.org/2440113002
Cr-Commit-Position: refs/heads/master@{#14762}
XGetImage() may return NULL and XServerPixelBuffer wasn't handling this
case properly.
BUG=649487
Review-Url: https://codereview.webrtc.org/2446733003
Cr-Commit-Position: refs/heads/master@{#14754}
It turns out that that audio network adaptor can always be created with knowledge of receiver frame length range. There is no need to keep some infrastructure that is used for runtime setting of receiver frame length ranges.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2429503002
Cr-Commit-Position: refs/heads/master@{#14748}
PacketList is now list<Packet> instead of list<Packet*>.
Splicing the lists in NetEqImpl::InsertPacketInternal instead of
moving packets. Avoid moving the packet when doing Rfc3389Cng.
Removed PacketBuffer::DeleteFirstPacket and DeleteAllPackets.
BUG=chromium:657300
Review-Url: https://codereview.webrtc.org/2425223002
Cr-Commit-Position: refs/heads/master@{#14747}
The mixer allocates an audio frame for each added data source. This
audio frame was deallocated when a source was removed from the
mixer. Source removal could happen during the mixing, and the existing
locking scheme (and the Clang thread checker) was not sufficient to
prevent a data race.
After this change, the mixer doesn't release its lock until it is
finished with the sources' Audio frames. Since multi-threaded access to
the mixer only happens when a source is added or removed, we believe
that this change wouldn't have any noticeable performance impact.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2439283002
Cr-Commit-Position: refs/heads/master@{#14744}
Separate the null implementation from rtp_rtcp_defines.h, and follow chromium style guide for virtual functions.
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2400993002
Cr-Commit-Position: refs/heads/master@{#14738}
the render side to the capture side using the same
pattern. Currently this is done independently for the
submodules.
This CL moves the the AEC functionality for this into
APM.
BUG=webrtc:5298, webrtc:6540
Review-Url: https://codereview.webrtc.org/2427553003
Cr-Commit-Position: refs/heads/master@{#14726}
Now does level estimate on the audio threads to avoid complex
copying of audio data to task queue. The old implementation could
also crash due to unclear ownership of the audio buffer.
BUG=webrtc:6569
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/2433393002 .
Cr-Commit-Position: refs/heads/master@{#14720}
Simplify the AudioMixer::Source interface and update the mixer
implementation to the new interface.
Instead of asking a mixer source to provide a pointer to an AudioFrame
during each mixing iteration, a mixer should supply a pointer to its
own AudioFrame.
This simplifies lifetime issues as sources do not give away an
internal pointer.
Implementation: when an audio source is added, the mixer allocates a
new AudioFrame. The audio frame is kept together in the internal class
SourceStatus together with the audio source pointer until the source
is removed.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2420913002
Cr-Commit-Position: refs/heads/master@{#14713}
This change is due to an incorrect understanding of the threading
model in Chrome. The new AudioMixer has a thread checker to ensure
that mixing is always done from a single thread. Mixing is done on the
Audio Output Thread. When run in Chrome, it can change. Even if the thread
changes, there is never more than one audio thread, and mixing is done
sequentially.
The threading checks and variable access checks are replaced with
rtc::RaceChecker counterparts.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2437913003
Cr-Commit-Position: refs/heads/master@{#14712}
Previously shared_desktop_frame.cc wasn't in primitives target, but it
is used in remoting client on android and the client should depend only
on primitives, but not desktop_capture overall.
BUG=653612
R=nicholss@chromium.org
Review URL: https://codereview.webrtc.org/2436913002 .
Cr-Commit-Position: refs/heads/master@{#14709}
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.
This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.
It will also create less build dependencies when the new mixer has replaced the old one.
NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
This CL makes scaling and cropping lazy in AVFoundationVideoCapturer and
provides optimized paths for SW and HW encoding. For SW encoding, an
efficient NV12 -> I420 cropping and scaling is implemented in
CoreVideoFrameBuffer::NativeToI420. For HW encoding, an efficient NV12 ->
NV12 cropping and scaling is implemented in
CoreVideoFrameBuffer::CropAndScaleTo. The performance improvement over
the existing cropping and scaling is that it is now done in one step
instead of making an intermediary copy of the Y plane.
There might still be room for improvement in the HW path using some HW
support. That will be explored in a future CL.
BUG=b/30939444
Review-Url: https://codereview.webrtc.org/2394483005
Cr-Commit-Position: refs/heads/master@{#14701}
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}
