Rename SetCodecAndOptions to SetCodec, it no longer sets or uses the
VideoOptions. In MediaConfig, collect the video-related flags into a
struct.
As a followup, it should be possible to delete VideoOptions from
VideoSendParameters and VideoSendStreamParameters.
TBR=pthatcher@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1745003002
Cr-Commit-Position: refs/heads/master@{#11828}
The current implementation is unnecessary expensive - we create a local reference frame for creating new Java objects and then create a new local reference. It's cheaper to just do jni->IsSameObject(obj, nullptr).
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1741723002 .
Cr-Commit-Position: refs/heads/master@{#11825}
For backwards compatibility, I've added kept the old interface to
Encode() and EncodeInternal and created default implementations of both
variants of EncodeInternal(), each calling the other. At least one of
the variants must be implemented in a subclass or we'll run out of stack
and explode. Would be nice if we could catch that before runtime. :/
The new interface to EncodeInternal() is protected, since it should
never be called from the outside.
Was unable to mark the old EncodeInternal() as RTC_DEPRECATED, since the
default implementaion of the new variant needs to call it to work around
old implementations. The old Encode() variant is deprecated, at least.
Added a test for backwards compatibility in audio_encoder_unittest.cc.
For the added test I broke out MockEncodeHelper from
audio_encoder_copy_red_unittest.cc and renamed it MockAudioEncoderHelper.
Review URL: https://codereview.webrtc.org/1725143003
Cr-Commit-Position: refs/heads/master@{#11823}
render_time time field (means capture time for sender side) is used by rtcp SenderReport to calculate offset since last frame and to estimate rtp timestamp for the time SenderReport should be send at.
mapping between rtp timestamp and ntp time in SenderReport is used for stream synchronization.
calculation of rtp_timestamp (using ntp_time of incoming video frame) for rtp packets is unchanged.
BUG=webrtc:5433, webrtc:5504, webrtc:5505
Review URL: https://codereview.webrtc.org/1693443002
Cr-Commit-Position: refs/heads/master@{#11820}
Chromium doesn't use the device managment code in webrtc/media
so we need a way to turn it off in order to eliminate Chromium's
src/third_party/libjingle/libjingle.gyp
BUG=webrtc:4256
NOTRY=True
TESTED=Trybots + successfully compiled with
GYP_DEFINES=include_internal_device_management=0 webrtc/build/gyp_webrtc
ninja -C out/Debug rtc_media
Review URL: https://codereview.webrtc.org/1693803002
Cr-Commit-Position: refs/heads/master@{#11816}
Permits measuring times from start of recording (usually start of a
call), and not time from first event that occurs after tracing starts.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1746693002 .
Cr-Commit-Position: refs/heads/master@{#11815}
The audio level of the AEC's output level was calculated before overlapping add, and therefore, a compensation was needed. The compensation is multiplying the level by 2 since, before overlapping add, the level is roughly halved due to windowing.
This had to be that way because the level was calculated in frequency domain and the signal after overlapping add has only its time domain representation.
The level calculation has been updated to work on time domain signal and therefore the problem is not there any longer.
This CL is to put the calculation of the AEC output level after overlapping add and remove the compensation.
BUG=
R=peah@webrtc.org
Review URL: https://codereview.webrtc.org/1644133002 .
Cr-Commit-Position: refs/heads/master@{#11810}
I readded virtual bool Pause(bool paused) for now with a dummy implementation since Chrome remoting override this method.
Original cl description:
Removed unused cricket::VideoCapturer methods:
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
bool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()
This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.
There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1744153002 .
Cr-Commit-Position: refs/heads/master@{#11809}
The purose of this cl is to remove dependency on cricket::VideoCapturer from WebRtcVideoChannel2.
This cl change CPU adaptation to use a new VideoSinkWants.Resolution
Cl is WIP and uploaded to start the discussion.
Tested on a N5 with hw acceleration turned off.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1695263002
Cr-Commit-Position: refs/heads/master@{#11804}
As a follow-up to https://codereview.webrtc.org/1737053006/ this CL further
improves memory usage by lazily allocating output buffers up to the passed
maximum size. This also changes the output buffer to a Buffer object.
BUG=
Review URL: https://codereview.webrtc.org/1741413002
Cr-Commit-Position: refs/heads/master@{#11801}
This change is done to remove abnormally high decode time measurements for H264 decoding. H264 decoding sometimes keeps a few frames as reference before outputting a new decoded frame. This pipeline causes some frames to get stuck when the source stops sending new frames. When the source starts sending frames again, the decode time measurements for the frames that were stuck will include the pause time, which can be arbitrary high. This CL is a simple fix for this problem by constraining the decode time values to a "reasonable" range.
BUG=b/27306053
Review URL: https://codereview.webrtc.org/1725243007
Cr-Commit-Position: refs/heads/master@{#11792}
Listening sockets will not read/write directly, so they don't need buffers.
BUG=
Review URL: https://codereview.webrtc.org/1737053006
Cr-Commit-Position: refs/heads/master@{#11791}
Currently, VideoCapturerAndroid owns a dedicated tread, and
SurfaceTextureHelper get this thread passed in the ctor. In
VideoCapturerAndroid.dispose(), ownership of the thread is passed to
SurfaceTextureHelper so that we can return directly instead of waiting
for the last frame to return.
This CL makes the SurfaceTextureHelper own the thread the whole time
instead, and VideoCapturerAndroid calls getHandler() to get it instead.
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1738123002
Cr-Commit-Position: refs/heads/master@{#11790}
Reason for revert:
Breaks downstream compilation. Please make non-breaking API changes for the reland or coordinate fixing downstream code quickly with the sheriff.
Original issue's description:
> Cleanup of webrtc::VideoFrame.
>
> Delete EqualsFrame method, used only by tests. Delete one of the
> CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
> and CopyFrame.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/208019637bfed975f8f13b16d40b90e200763cd6
> Cr-Commit-Position: refs/heads/master@{#11783}
TBR=pbos@webrtc.org,perkj@webrtc.org,pthatcher@webrtc.org,mflodman@webrtc.org,marpan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1743613002
Cr-Commit-Position: refs/heads/master@{#11789}
Removes per-extension functions in ViEChannel/ViEReceiver and instead
register extensions directly on the RTP module by mapping extension
string to RTP-header-extension type.
BUG=webrtc:5494
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1740133002 .
Cr-Commit-Position: refs/heads/master@{#11786}
Delete EqualsFrame method, used only by tests. Delete one of the
CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
and CopyFrame.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1679323002
Cr-Commit-Position: refs/heads/master@{#11783}
Reason for revert:
Breaks remoting::protocol::WebrtcVideoCapturerAdapter::Pause'
See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3689/steps/compile/logs/stdio
Original issue's description:
> Removed unused cricket::VideoCapturer methods:
>
> void UpdateAspectRatio(int ratio_w, int ratio_h);
> void ClearAspectRatio();
> ool Pause(bool paused);
> Restart(const VideoFormat& capture_format);
> MuteToBlackThenPause(bool muted);
> IsMuted() const
> set_square_pixel_aspect_ratio
> bool square_pixel_aspect_ratio()
>
> This cl also remove the use of messages and posting of state change.
> Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
> It does not add restrictions on what thread frames are delivered on though.
>
> There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/e9c0cdff2dad2553b6ff6820c0c7429cb2854861
> Cr-Commit-Position: refs/heads/master@{#11773}
TBR=magjed@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1740963002
Cr-Commit-Position: refs/heads/master@{#11777}
By doing an unsigned instead of a signed addition, we get the exact
same machine code (in non-UBSan builds), but no longer trigger
undefined behavior since unsigned overflow is defined behavior.
BUG=webrtc:5485
Review URL: https://codereview.webrtc.org/1734883003
Cr-Commit-Position: refs/heads/master@{#11776}
This patch tries to only change the interface to VideoTrack, with
minimal changes to the implementation. Some points worth noting:
VideoTrackRenderers should ultimately be deleted, but it is kept for
now since we need an object implementing webrtc::VideoRenderer, and
that shouldn't be VideoTrack.
BUG=webrtc:5426
TBR=glaznev@webrtc.org // please look at examples
Review URL: https://codereview.webrtc.org/1684423002
Cr-Commit-Position: refs/heads/master@{#11775}
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
ool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()
This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.
There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1733673002
Cr-Commit-Position: refs/heads/master@{#11773}
Code still compiles in Chromium with a proper const float* variable so
it is expected to address the issue.
BUG=chromium:589951
TBR=peah@webrtc.org
Review URL: https://codereview.webrtc.org/1739893004 .
Cr-Commit-Position: refs/heads/master@{#11772}
Instead relies on SetSendingMediaStatus() to filter out receiving RTP
modules. This status is now set in VoiceEngine's SetSend() for senders
along with SetSendingStatus().
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1705763002 .
Cr-Commit-Position: refs/heads/master@{#11768}
This CL introduces a new class CopyOnWriteBuffer that holds data in a
refcounted Buffer which is shared between copied CopyOnWriteBuffer to avoid
unnecessary allocations / memory copies.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1697743003
Cr-Commit-Position: refs/heads/master@{#11767}
