This will prevent it from blocking network input when it falls behind,
which is happening when running with ThreadSanitizer.
BUG=webrtc:4663
Review URL: https://codereview.webrtc.org/1236023010
Cr-Commit-Position: refs/heads/master@{#9707}
Reason for revert:
AppRTCDemo often crashes in loopback mode and incorrect layout when connection is established
BUG=webrtc:4909,webrtc:4910
Original issue's description:
> AppRTCDemo: Render each video in a separate SurfaceView
>
> This CL introduces a new org.webrtc.VideoRenderer.Callbacks implementation called SurfaceViewRenderer that renders each video stream in its own SurfaceView. AppRTCDemo is updated to use this new rendering.
>
> This CL also does the following changes:
> * Make the VideoRenderer.Callbacks interface asynchronous and require that renderFrameDone() is called for every renderFrame(). In JNI, this is implemented with cricket::VideoFrame::Copy()/delete.
> * Make public static helper functions: convertScalingTypeToVisibleFraction(), getDisplaySize(), and getTextureMatrix().
> * Introduces new helper functions surfaceWidth()/surfaceHeight() in EGlBase that allows to query the surface size.
> * Introduce PercentFrameLayout that implements the percentage layout that is used by AppRTCDemo.
>
> BUG=webrtc:4742
>
> Committed: https://crrev.com/05bfbe47ef6bcc9ca731c0fa0d5cd15a4f21e93f
> Cr-Commit-Position: refs/heads/master@{#9699}
TBR=glaznev@webrtc.org,wzh@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1286133002
Cr-Commit-Position: refs/heads/master@{#9703}
This CL introduces a new org.webrtc.VideoRenderer.Callbacks implementation called SurfaceViewRenderer that renders each video stream in its own SurfaceView. AppRTCDemo is updated to use this new rendering.
This CL also does the following changes:
* Make the VideoRenderer.Callbacks interface asynchronous and require that renderFrameDone() is called for every renderFrame(). In JNI, this is implemented with cricket::VideoFrame::Copy()/delete.
* Make public static helper functions: convertScalingTypeToVisibleFraction(), getDisplaySize(), and getTextureMatrix().
* Introduces new helper functions surfaceWidth()/surfaceHeight() in EGlBase that allows to query the surface size.
* Introduce PercentFrameLayout that implements the percentage layout that is used by AppRTCDemo.
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1257043004
Cr-Commit-Position: refs/heads/master@{#9699}
DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002
The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.
Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.
For more information about the steps being taken to land this without breaking Chromium, see referenced bug.
BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1176383004 .
Cr-Commit-Position: refs/heads/master@{#9696}
This CL makes sure the methods are always called on the correct thread.
Review URL: https://codereview.webrtc.org/1235263003
Cr-Commit-Position: refs/heads/master@{#9688}
Currently, we only return frames if CreateAliasedFrame() is called, which is not the case for dropped frames.
Review URL: https://codereview.webrtc.org/1268333005
Cr-Commit-Position: refs/heads/master@{#9683}
Significant changes:
- move the libjingle_examples.gyp file into webrtc directory.
- rename talk/examples/android to webrtc/examples/androidapp to avoid name conflicts.
- update paths in talk/libjingle_tests.gyp to point to webrtc directory for Objective-C test.
BUG=
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1235563006 .
Cr-Commit-Position: refs/heads/master@{#9681}
New PeerConnectionFactoryInterface::CreatePeerConnection taking both service and store added (old CreatePC signature still exists).
This is CL is part of an effort to land https://codereview.webrtc.org/1176383004 without breaking Chromium.
See bug for more information.
BUG=webrtc:4899
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1268363002 .
Cr-Commit-Position: refs/heads/master@{#9680}
onPreviewFrame() might be called with a null data pointer, which is allowed according to the documentation.
BUG=webrtc:4877
Review URL: https://codereview.webrtc.org/1260183004
Cr-Commit-Position: refs/heads/master@{#9674}
For now add only Galaxy S4 to the list, since its H.264 HW encoder
generates two times lower bitrate comparing to target.
Also use VBR mode for H.264 encoder configuration.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1270603007 .
Cr-Commit-Position: refs/heads/master@{#9673}
Permits setting RTP extensions for AudioReceiveStream without enabling
combined A/V BWE. This prevents spamming the log with "Failed to find
extension id:".
BUG=webrtc:4870
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1256803004
Cr-Commit-Position: refs/heads/master@{#9633}
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
An earlier version of this commit caused a crash on AEC dump.
TBR=aluebs@webrtc.org,pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1248393003 .
Cr-Commit-Position: refs/heads/master@{#9626}
Reason for revert:
I think this causes WebRtcBrowserTest.CallAndModifyStream to fail on Android. See https://code.google.com/p/webrtc/issues/detail?id=4857 for more info.
Original issue's description:
> Fixing scenario where track is rejected and later un-rejected.
>
> Added `RestartLocalTracks` and `RestartRemoteTracks` methods to
> `MediaStreamHandlerContainer` which will redo the track handlers'
> initial setup; most importantly, this will re-connect the
> renderer/capturer/etc. to a channel which was destroyed and then
> re-created.
>
> Also added `AcceptRemoteTracks` method to MediaStreamSignaling, which
> does the inverse of `RejectRemoteTracks`. Effectively this will notify
> sinks that the track is live again, after previously being set to
> `kEnded` when it was rejected.
>
> BUG=webrtc:2136
>
> Committed: https://crrev.com/be37888b6d5d269dbd5385569dba15c0d70594f2
> Cr-Commit-Position: refs/heads/master@{#9600}
TBR=pthatcher@webrtc.org,juberti@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1247443005
Cr-Commit-Position: refs/heads/master@{#9622}
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388
Sample output:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib: extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[ FAILED ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam = and GetParam() = (361 ms)
Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: c204754b7aTBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1253573005
Cr-Commit-Position: refs/heads/master@{#9621}
Added `RestartLocalTracks` and `RestartRemoteTracks` methods to
`MediaStreamHandlerContainer` which will redo the track handlers'
initial setup; most importantly, this will re-connect the
renderer/capturer/etc. to a channel which was destroyed and then
re-created.
Also added `AcceptRemoteTracks` method to MediaStreamSignaling, which
does the inverse of `RejectRemoteTracks`. Effectively this will notify
sinks that the track is live again, after previously being set to
`kEnded` when it was rejected.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1231613002
Cr-Commit-Position: refs/heads/master@{#9600}
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.
Review URL: https://codereview.webrtc.org/1225153002
Cr-Commit-Position: refs/heads/master@{#9597}
"field_trial::FindFullName" can return "std::string()" which should not
be referenced by the caller.
Review URL: https://codereview.webrtc.org/1238943003
Cr-Commit-Position: refs/heads/master@{#9594}
This CL improves the memory footprint a bit by using string references
instead of creating a copy.
Review URL: https://codereview.webrtc.org/1241973002
Cr-Commit-Position: refs/heads/master@{#9592}
BUG=webrtc:4690
Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1226123005 .
Cr-Commit-Position: refs/heads/master@{#9591}
Tested that this doesn't break compatibility with Firefox or older
versions of Chrome, no matter which side generates the initial offer.
BUG=webrtc:2796
Review URL: https://codereview.webrtc.org/1219333002
Cr-Commit-Position: refs/heads/master@{#9589}
2. provide an implementation for SetIceConnectionReceivingTimeout so that Chrome does not complain.
BUG=
Review URL: https://codereview.webrtc.org/1227843006
Cr-Commit-Position: refs/heads/master@{#9574}
Allows removing MediaRecorder which isn't in use apart from channel
unittests, along with it unittests for MediaRecorder that are flaky when
run in parallel can also go.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1219663008
Cr-Commit-Position: refs/heads/master@{#9558}
This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.
BUG=chromium:507307
Review URL: https://codereview.webrtc.org/1231823002
Cr-Commit-Position: refs/heads/master@{#9557}
