8673 Commits

Author SHA1 Message Date
Sergey Ulanov
58000a0c3d Move shared_desktop_frame.cc to webrtc/modules/desktop_capture:primitives
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}
2016-10-20 16:34:04 +00:00
danilchap
142f019d87 Append second nack list in same compound rtcp packet
instead of replace

BUG=webrtc:6483

Review-Url: https://codereview.webrtc.org/2426543002
Cr-Commit-Position: refs/heads/master@{#14708}
2016-10-20 15:22:45 +00:00
aleloi
aed581a4f3 Made AudioReceiveStream a mixer participant.
Methods to facilitate this are added to ChannelProxy and voe::Channel.

BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2378143004
Cr-Commit-Position: refs/heads/master@{#14707}
2016-10-20 13:32:47 +00:00
sakal
5f70d3b559 Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest.
ArgumentMatchers is the new name of the class. Also, we don't need to
pass a class to the isNotNull matcher.

BUG=webrtc:6558
R=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2440433002
Cr-Commit-Position: refs/heads/master@{#14706}
2016-10-20 12:49:12 +00:00
aleloi
201dfe90a7 Split audio mixer into interface and implementation.
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}
2016-10-20 12:06:44 +00:00
brandtr
76648da8dc Add FlexfecReceiveStream.
This class is logically parallel with the {Audio,Video}ReceiveStream
classes. Its purpose is to describe a receive stream of FlexFEC packets,
through the corresponding config.

Functionally, this class simply forwards the received RTP packets
to its FlexfecReceiver, which returns recovered packets to the
Call level, for appropriate demultiplexing based on SSRC.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2397843005
Cr-Commit-Position: refs/heads/master@{#14704}
2016-10-20 11:54:51 +00:00
Henrik Kjellander
057b8d91a6 Remove all traces of Dr Memory.
The Dr Memory toolchain is no longer supported by Chromium and
their bots have been removed. WebRTC will now rely on the LLVM
santizers for catching such errors.

BUG=webrtc:6553
NOTRY=True
R=ehmaldonado@webrtc.org

Review URL: https://codereview.webrtc.org/2434563003 .

Cr-Commit-Position: refs/heads/master@{#14703}
2016-10-20 11:17:25 +00:00
palmkvist
69034df0b0 Make GN build screenshare_loopback
BUG=None

Review-Url: https://codereview.webrtc.org/2338233004
Cr-Commit-Position: refs/heads/master@{#14702}
2016-10-20 11:03:32 +00:00
magjed
5a8724564c iOS: Optimize video scaling and cropping
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}
2016-10-20 10:34:32 +00:00
minyue
7a973447eb Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
BUG=webrtc:5806, webrtc:4690

Review-Url: https://codereview.webrtc.org/2405183002
Cr-Commit-Position: refs/heads/master@{#14700}
2016-10-20 10:27:21 +00:00
magjed
1cb48232ac Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer
This CL changes YuvConverter to use an OpenGL Framebuffer as rendering
target instead of an EGL pixel buffer surface. The purpose is to reduce
the number of EGL contexts and to be able to use YuvConverter from
EglRenderer without having to detach the EGL surface.

BUG=webrtc:6470

Review-Url: https://codereview.webrtc.org/2436653003
Cr-Commit-Position: refs/heads/master@{#14699}
2016-10-20 10:19:20 +00:00
magjed
9ab8a1884d Android: Extend functionality of EglRenderer
The purpose is to prepare for a TextureViewRenderer that will need the
new functionality.

The new functionality is:
 * Be able to create an EglRenderer using a SurfaceTexture.
 * Fps reduction logic.
 * Log statistics every 4 seconds regardless of framerate.
 * Include swap buffer time in statistics.
 * Use EglBase10 if texture frames are disabled.
 * Function for printing stack trace of render thread.
 * Public clearImage() function for clearing the EGLSurface.

BUG=webrtc:6470

Review-Url: https://codereview.webrtc.org/2428933002
Cr-Commit-Position: refs/heads/master@{#14698}
2016-10-20 10:18:15 +00:00
nisse
ca20e7cfed Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ )
Reason for revert:
Turned out this broke a downstream app. Reverting while investigating.

Original issue's description:
> Delete unused file mediacommon.h.
>
> BUG=None
>
> Committed: https://crrev.com/a34e796a1b59a1068c77a3e36d2a24fd37c08afc
> Cr-Commit-Position: refs/heads/master@{#14689}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2441493003
Cr-Commit-Position: refs/heads/master@{#14697}
2016-10-20 09:53:06 +00:00
sakal
c1f8ecb661 Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient.
Since the class no longer creates its own videoCapturer and it may not
even be a camera, the check doesn't make sense anymore. This also
removes some of the build warnings.

BUG=webrtc:6558

Review-Url: https://codereview.webrtc.org/2434533003
Cr-Commit-Position: refs/heads/master@{#14696}
2016-10-20 09:34:49 +00:00
sakal
be4aff76a6 Suppress deprecation warning in CallFragment.
The old signature of this method has been deprecated. However, the new
signature is only support API level 23+, so we can't really use it yet.

BUG=webrtc:6558

Review-Url: https://codereview.webrtc.org/2437593004
Cr-Commit-Position: refs/heads/master@{#14695}
2016-10-20 09:32:53 +00:00
aleloi
e33c5d918a Added a level controller initialization value to MediaConstraints.
An audio track with a level controller with the correct initialization
value can be created by a combination of
PeerConnectionFactory::CreateAudioTrack(..., audio_source) and
either
audio_source = PeerConnectionFactory::CreateAudioSource(constraints) or
audio_source = PeerConnectionFactory::CreateAudioSource(audio_options).

NOTRY=True
BUG=webrtc:6386

Review-Url: https://codereview.webrtc.org/2408143003
Cr-Commit-Position: refs/heads/master@{#14693}
2016-10-20 08:53:30 +00:00
denicija
647915fb95 Add loopback option and improve UX of AppRTCMobile for Mac.
CL introduces changes in the layout of the mac demo app.
Bellow are the listed changes
* The label above the room field is removed and that text is shown in the text field's placeholder.
* Size of the window is decreased to accomodate smaller resolutions
* The local view is shown on top of remote view in 3/rd of it's size
* The log view is bellow the remote view and shows client status messages as well as some simple usage instructions.
* Loopback functionality is added
* Start call button added for improved usability

Note: This file was not following our internal ObjC style guidelines i.e
NSString *a - right
NSString* a - wrong
However I've decided to submit the changes and to follow the current style.
I'll submit a bug to update the style in separate CL

BUG=webrtc:6496

Review-Url: https://codereview.webrtc.org/2396333002
Cr-Commit-Position: refs/heads/master@{#14692}
2016-10-20 08:46:49 +00:00
kthelgason
725e2121ad Prevent stripping of C interfaces in framework
We need to redaclare certain interfaces in the GN build files for the
macOS and iOS framework targets to prevent those symbols being stripped
by the linker.

BUG=webrtc:6434

Review-Url: https://codereview.webrtc.org/2431603002
Cr-Commit-Position: refs/heads/master@{#14691}
2016-10-20 08:31:48 +00:00
danilchap
e037060161 Add to rtc::Optional equality/unequality comparisions with object
BUG=None

Review-Url: https://codereview.webrtc.org/2432393002
Cr-Commit-Position: refs/heads/master@{#14690}
2016-10-20 07:58:31 +00:00
nisse
a34e796a1b Delete unused file mediacommon.h.
BUG=None

Review-Url: https://codereview.webrtc.org/2437703002
Cr-Commit-Position: refs/heads/master@{#14689}
2016-10-20 07:45:02 +00:00
kthelgason
55928fef1e QualityScaler reset bugfix
BUG=webrtc:6563
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2434803002
Cr-Commit-Position: refs/heads/master@{#14688}
2016-10-20 07:42:59 +00:00
perkj
0489e498eb Change RefCountedObject to use perfect forwarding.
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}
2016-10-20 07:24:06 +00:00
zijiehe
79f0bf3ab5 A variable in ScreenCapturerWinDirectx has a bad name
BUG=https://bugs.chromium.org/p/chromium/issues/detail?id=314516

Review-Url: https://codereview.webrtc.org/2440613002
Cr-Commit-Position: refs/heads/master@{#14686}
2016-10-20 06:40:44 +00:00
kthelgason
f04f14e772 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

Review-Url: https://codereview.webrtc.org/2430353004
Cr-Commit-Position: refs/heads/master@{#14685}
2016-10-19 17:34:39 +00:00
kthelgason
cc6817e9ce 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

Review-Url: https://codereview.webrtc.org/2370853005
Cr-Commit-Position: refs/heads/master@{#14684}
2016-10-19 16:31:15 +00:00
Magnus Jedvert
577bc19210 Android: Move YuvConverter to its own file
YuvConverter is complex class that deserves its own file. It is also used outside of SurfaceTextureHelper.

BUG=webrtc:6470
R=sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2426023002 .

Cr-Commit-Position: refs/heads/master@{#14683}
2016-10-19 13:29:13 +00:00
danilchap
b6f1fb5337 Delete RTPSender::BuildRtpHeader function
and all dependencies

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2399463009
Cr-Commit-Position: refs/heads/master@{#14682}
2016-10-19 13:11:44 +00:00
Per
061ea0df03 Remove VideoCodec resolution validation.
This is needed to be able to land https://codereview.webrtc.org/2408153002/  "Remove cricket::VideoCodec with, height and framerate properties" without breaking upstream projects.

BUG=webrtc:5332
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2437453003 .

Cr-Commit-Position: refs/heads/master@{#14681}
2016-10-19 11:58:09 +00:00
sakal
73c5d4a083 Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar
This makes it possible for external applications to use this class.

BUG=webrtc:6524
NOTRY=True

Review-Url: https://codereview.webrtc.org/2430693002
Cr-Commit-Position: refs/heads/master@{#14679}
2016-10-19 09:46:27 +00:00
nisse
09347858f7 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ )
This cl now makes cricket::VideoFrame and cricket::WebRtcVideoFrame aliases for webrtc::VideoFrame.

Reason for revert:
Fixing backwards compatibility issues.

Original issue's description:
> Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ )
>
> Reason for revert:
> Breaks compile for Chromium builds:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/10761
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/18142
>
> FAILED: obj/remoting/protocol/protocol/webrtc_video_renderer_adapter.o
> ../../remoting/protocol/webrtc_video_renderer_adapter.cc:110:52: error: no member named 'transport_frame_id' in 'cricket::VideoFrame'
>                  weak_factory_.GetWeakPtr(), frame.transport_frame_id(),
>                                              ~~~~~ ^
> 1 error generated.
>
> Please run chromium trybots as described at https://webrtc.org/contributing/#tryjobs-on-chromium-trybots before relanding.
>
> Original issue's description:
> > Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
> > all methods but a few constructors. And similarly for the
> > subclass cricket::WebRtcVideoFrame.
> >
> > TBR=tkchin@webrtc.org  # Added an include line
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/dda6ec008a0fc8d52e118814fb779032e8931968
> > Cr-Commit-Position: refs/heads/master@{#14576}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,nisse@webrtc.org
> NOTRY=True
> NOPRESUBMIT=True
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d36dd499c8f253cbcf37364c2a070c2e8c7100e9
> Cr-Commit-Position: refs/heads/master@{#14583}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2411953002
Cr-Commit-Position: refs/heads/master@{#14678}
2016-10-19 07:30:35 +00:00
brandtr
4e52386339 Reland of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #1 id:1 of https://codereview.webrtc.org/2427733002/ )
Reason for revert:
Flaky test has been fixed.

Original issue's description:
> Revert of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #2 id:60001 of https://codereview.webrtc.org/2390823009/ )
>
> Reason for revert:
> Speculative revert as it may be the cause of the DrMemory test failure:
> https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/5115
>
> Original issue's description:
> > Add path for recovered packets from internal::Call to RtpStreamReceiver.
> >
> > When the FlexfecReceiver recovers media packets, it inserts these into
> > internal::Call, which then distributes them to the appropriate
> > VideoReceiveStream/RtpStreamReceiver.
> >
> > BUG=webrtc:5654
> >
> > Committed: https://crrev.com/9c4b4b47f4325b48e1856566a30983f9e4e30dd0
> > Cr-Commit-Position: refs/heads/master@{#14642}
>
> TBR=stefan@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5654
>
> Committed: https://crrev.com/862d74d0176fa762b3c96cf20bd36f27e7001a47
> Cr-Commit-Position: refs/heads/master@{#14652}

TBR=stefan@webrtc.org,honghaiz@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2428303004
Cr-Commit-Position: refs/heads/master@{#14677}
2016-10-19 06:50:53 +00:00
zijiehe
249beee124 Remove DesktopRegion parameter from DesktopCapturer::Capture
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=6513

Review-Url: https://codereview.webrtc.org/2433503002
Cr-Commit-Position: refs/heads/master@{#14676}
2016-10-19 06:13:38 +00:00
zijiehe
6a4607e100 Deflaky ScreenCapturerTest
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}
2016-10-19 01:22:25 +00:00
Irfan Sheriff
1eb12934e7 Handle BW drop in ALR region and initiate probing
Original change by isheriff@chromium.org: http://crrev.com/2387463002#ps40001

BUG=webrtc:6332
TBR=philipel@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2432633002 .

Patch from Irfan Sheriff <isheriff@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#14673}
2016-10-19 00:04:35 +00:00
johan
a9c7cfac41 Prepare for introduction of rtc::PacketTransportInterface.
A rtc::PacketTransportInterface typedef is introduced to allow preparing
downstream projects for the upcoming refactoring of
cricket::Transport. This refactoring will introduce
rtc::PacketTransportInterface in https://codereview.webrtc.org/2416023002/ .

BUG=webrtc:6531

Review-Url: https://codereview.webrtc.org/2429803002
Cr-Commit-Position: refs/heads/master@{#14672}
2016-10-18 22:38:43 +00:00
bertholdherrmann08
1203066236 Compilerwarning possible loss of data in file port.h
BUG=webrtc:6179

Review-Url: https://codereview.webrtc.org/2224323002
Cr-Commit-Position: refs/heads/master@{#14671}
2016-10-18 21:00:06 +00:00
hbos
cc555c5019 RTCDataChannelStats[1] added, supporting all stats members.
Also updates MockDataChannel to also mock id, messages_sent, bytes_sent,
messages_received and bytes_received.

[1] https://w3c.github.io/webrtc-stats/#dcstats-dict*

BUG=chromium:654927, chromium:627816

Review-Url: https://codereview.webrtc.org/2420473002
Cr-Commit-Position: refs/heads/master@{#14670}
2016-10-18 19:48:37 +00:00
asapersson
1394c7b594 Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx
Add a limit for minimum number of frames to be received before verifying histograms stats to reduce flakyness.

BUG=webrtc:6509

Review-Url: https://codereview.webrtc.org/2420443002
Cr-Commit-Position: refs/heads/master@{#14669}
2016-10-18 18:50:57 +00:00
michaelt
9960bb1469 Call OnTransportFeedback just when feedback_observer exist.
BUG=webrtc:6523

Review-Url: https://codereview.webrtc.org/2404233004
Cr-Commit-Position: refs/heads/master@{#14667}
2016-10-18 16:40:38 +00:00
michaelt
53fe19d6f3 Set min and max rate on caller and on callee side.
BUG=webrtc:6518

Review-Url: https://codereview.webrtc.org/2410903002
Cr-Commit-Position: refs/heads/master@{#14666}
2016-10-18 16:39:28 +00:00
mandermo
64e1a32e2f Second try to get "Support for video file instead of camera and output video out to file" accepted
The old CL can be found here: https://codereview.webrtc.org/2273573003/

The orginal broke down stream, this CL tries to solve those issues.

BUG=webrtc:6545

Review-Url: https://codereview.webrtc.org/2426003002
Cr-Commit-Position: refs/heads/master@{#14665}
2016-10-18 15:47:59 +00:00
kjellander
67a8c986ab Revert of Support for video file instead of camera and output video out to file (patchset #17 id:320001 of https://codereview.webrtc.org/2273573003/ )
Reason for revert:
Breaks internal project.

Original issue's description:
> Support for video file instead of camera and output video out to file
>
> When video out to file is enabled the remote video which is recorded is
> not show on screen.
>
> You can use this command line for file input and output:
> monkeyrunner ./webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py --devname 02157df28cd47001 --videoin /storage/emulated/0/reference_video_1280x720_30fps.y4m --videoout /storage/emulated/0/output.y4m --videoout_width 1280 --videoout_height 720 --videooutsave /tmp/out.y4m
>
> BUG=webrtc:6545
>
> Committed: https://crrev.com/44666997ca912705f8f96c9bd211e719525a3ccc
> Cr-Commit-Position: refs/heads/master@{#14660}

TBR=magjed@webrtc.org,sakal@webrtc.org,jansson@chromium.org,mandermo@google.com,mandermo@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6545

Review-Url: https://codereview.webrtc.org/2425763003
Cr-Commit-Position: refs/heads/master@{#14664}
2016-10-18 13:05:40 +00:00
nisse
f33970b15e Add unittest for I420Buffer::Rotate.
Also change pointer to const ref for CheckCrop helper.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2406463002
Cr-Commit-Position: refs/heads/master@{#14663}
2016-10-18 13:01:38 +00:00
terelius
6ed592d0ab Rename variables to reflect that DelayBasedBwe lives on the send side rather than receive side.
BUG=0

Review-Url: https://codereview.webrtc.org/2404253004
Cr-Commit-Position: refs/heads/master@{#14662}
2016-10-18 12:55:36 +00:00
henrika
5588a13fe7 Now uses rtc::Buffer in AudioDeviceBuffer.
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}
2016-10-18 12:14:35 +00:00
mandermo
44666997ca Support for video file instead of camera and output video out to file
When video out to file is enabled the remote video which is recorded is
not show on screen.

You can use this command line for file input and output:
monkeyrunner ./webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py --devname 02157df28cd47001 --videoin /storage/emulated/0/reference_video_1280x720_30fps.y4m --videoout /storage/emulated/0/output.y4m --videoout_width 1280 --videoout_height 720 --videooutsave /tmp/out.y4m

BUG=webrtc:6545

Review-Url: https://codereview.webrtc.org/2273573003
Cr-Commit-Position: refs/heads/master@{#14660}
2016-10-18 11:52:06 +00:00
danilchap
9e83c97e9f Add rtc::Optional::emplace
BUG=None

Review-Url: https://codereview.webrtc.org/2424063002
Cr-Commit-Position: refs/heads/master@{#14659}
2016-10-18 11:07:25 +00:00
ossu
7a3776102f Removed RTPHeader from NetEq's Packet struct.
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}
2016-10-18 11:06:19 +00:00
aleloi
553024ab34 During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
This change trims the potentially longer array to the size of the smaller one.

NOTRY=True
BUG=none

Review-Url: https://codereview.webrtc.org/2357883002
Cr-Commit-Position: refs/heads/master@{#14657}
2016-10-18 08:44:50 +00:00
brandtr
e405d9b8df Add a fuzzer for FlexfecReceiver.
Specifically set max_len to 2000, to simulate multi-packet insertions.

BUG=webrtc:5654
NOTRY=true

Review-Url: https://codereview.webrtc.org/2391263002
Cr-Commit-Position: refs/heads/master@{#14656}
2016-10-18 08:18:12 +00:00