402 Commits

Author SHA1 Message Date
kjellander
a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00
glaznev
ae95ff32ff Add more logging and fix PTS overflow for HW decoder.
- Reduce maximum pending frames for H.264 decoder to 8.
- Log data for next 2 output frames every time frame drop
happens or decoder drain is triggered.
- When timeout happens for dequeueInputBuffer call try to
drain the decoder and get input buffer one more time.
- Fix PTS values overflow.

Review URL: https://codereview.webrtc.org/1661203002

Cr-Commit-Position: refs/heads/master@{#11492}
2016-02-04 19:47:20 +00:00
honghaiz
9031d6366f Remove the network with empty name or NONE connection type from the network list.
In some device (e.g. Galaxy s6), the OS returns a list of network containing
one that has empty network name or NONE connection type, which cannot be used and cause crash to the app.

BUG=

Review URL: https://codereview.webrtc.org/1655313005

Cr-Commit-Position: refs/heads/master@{#11482}
2016-02-04 05:45:28 +00:00
Honghai Zhang
14d024d882 Do not notify networkconnect if the connection type is known.
This sometimes happened with sim card has a voice plan but does not have data plan.
Renable the DCHECK.

BUG=
R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1668673003 .

Cr-Commit-Position: refs/heads/master@{#11479}
2016-02-03 23:12:33 +00:00
Honghai Zhang
45b683f43f Call static method getConnectionType using the class name.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1669573002 .

Cr-Commit-Position: refs/heads/master@{#11478}
2016-02-03 22:15:12 +00:00
honghaiz
27a348555a Fixing a DCHECK failure on unknown connection type from OS.
Sometimes Android OS provides unknown connection type,
causing a DCHECK failure. This CL temporarily removes that checking.

BUG=
TBR=glaznev@webrtc.org
NOTRY=true

Review URL: https://codereview.webrtc.org/1667503002

Cr-Commit-Position: refs/heads/master@{#11466}
2016-02-03 02:20:31 +00:00
honghaiz
a7ad7c3ca0 Get the adapter type information from Android OS.
BUG=

Review URL: https://codereview.webrtc.org/1594673002

Cr-Commit-Position: refs/heads/master@{#11463}
2016-02-02 20:54:28 +00:00
Peter Boström
ed3277bf14 Deprecate VideoDecoder::Reset() and remove calls.
Removes calls to decoder reset and instead drops delta frames and
requests keyframes until one arrives.

BUG=webrtc:5475
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1647163002 .

Cr-Commit-Position: refs/heads/master@{#11460}
2016-02-02 14:40:13 +00:00
glaznev
94291480b6 Extra logging for HW codec.
- Add extra logging for Android HW codec corner cases
when frames are dropped or resolution is changed.
- Use presentation timestamps for decoded frame logging.
- Enable key frame sending on long frame gap for
H.264 codec.

BUG=b/26504665

Review URL: https://codereview.webrtc.org/1653523003

Cr-Commit-Position: refs/heads/master@{#11452}
2016-02-01 21:17:28 +00:00
Alex Glaznev
eee86a6aa3 Add option to disable particular HW video codec from app.
Plus minor clean up / adding comments.

BUG=b/26695339
R=jiayl@webrtc.org

Review URL: https://codereview.webrtc.org/1644253003 .

Cr-Commit-Position: refs/heads/master@{#11431}
2016-01-29 22:17:16 +00:00
terelius
6043f2e5d6 Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ )
Reason for revert:
onFirstMediaPacketReceived() breaks bot.

Original issue's description:
> Adding "first packet received" notification to PeerConnectionObserver.
>
> R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
>
> Committed: https://crrev.com/42265a8cc3b3f3db4aa2c29005aed2fb4393adef
> Cr-Commit-Position: refs/heads/master@{#11401}
>
> Committed: https://crrev.com/08a6eab75e13613183509d91d3892c1db57f6fc5
> Cr-Commit-Position: refs/heads/master@{#11404}

TBR=pthatcher@webrtc.org,tkchin@webrtc.org,glaznev@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1647483004

Cr-Commit-Position: refs/heads/master@{#11415}
2016-01-28 13:06:16 +00:00
glaznev
919ff75376 Use high QP threshold for HW VP8 encoder frame downscaling.
Before HW VP8 downscaling was triggered by frame drops only.
Also reset the encoder when it drops large amount of frames in a row.

BUG=b/26504665

Review URL: https://codereview.webrtc.org/1592883004

Cr-Commit-Position: refs/heads/master@{#11406}
2016-01-27 23:01:08 +00:00
Taylor Brandstetter
08a6eab75e Adding "first packet received" notification to PeerConnectionObserver.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Committed: https://crrev.com/42265a8cc3b3f3db4aa2c29005aed2fb4393adef
Cr-Commit-Position: refs/heads/master@{#11401}

Review URL: https://codereview.webrtc.org/1581693006 .

Cr-Commit-Position: refs/heads/master@{#11404}
2016-01-27 21:38:57 +00:00
deadbeef
7b3c72ffa9 Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #4 id:60001 of https://codereview.webrtc.org/1581693006/ )
Reason for revert:
Seems that the end-to-end unit tests are now flaky: https://build.chromium.org/p/client.webrtc/builders/Win64%20Debug/builds/6283

Will reland after fixing the test flakiness.

Original issue's description:
> Adding "first packet received" notification to PeerConnectionObserver.
>
> R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
>
> Committed: https://crrev.com/42265a8cc3b3f3db4aa2c29005aed2fb4393adef
> Cr-Commit-Position: refs/heads/master@{#11401}

TBR=pthatcher@webrtc.org,tkchin@webrtc.org,glaznev@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1640173004

Cr-Commit-Position: refs/heads/master@{#11402}
2016-01-27 21:03:47 +00:00
Taylor Brandstetter
42265a8cc3 Adding "first packet received" notification to PeerConnectionObserver.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1581693006 .

Cr-Commit-Position: refs/heads/master@{#11401}
2016-01-27 20:10:44 +00:00
Per
ec2922f864 Change PeerConnectionFactory.setVideoHwAccelerationOptions to create shared Egl context for harware encoders and decoders.
Before this fix, it was required that the EGL context used as an argument was kept open until all PeerConnections using the context had been closed. With this fix, that is no longer required.
Also, if a released EGLContext (EGL_NO_CONTEXT) is used, harware codecs will fallback to use byte buffers for encoding and decoding.
BUG=b/26583522
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1615153002 .

Cr-Commit-Position: refs/heads/master@{#11398}
2016-01-27 14:25:56 +00:00
noahric
5082c83320 Make type and constructors in EglBase14 public.
This type is used for GL contexts in android hardware encode/decode, so
it seems correct/useful to make it publicly constructable.

Review URL: https://codereview.webrtc.org/1604403002

Cr-Commit-Position: refs/heads/master@{#11331}
2016-01-21 10:13:00 +00:00
asapersson
1d61a518ac Send key frame if time difference between incoming frames exceeds a certain limit.
BUG=

Review URL: https://codereview.webrtc.org/1510913007

Cr-Commit-Position: refs/heads/master@{#11315}
2016-01-20 09:13:51 +00:00
nisse
c4c8485662 Deleted renderer-related SetSize methods, and all uses.
webrtc::VideoRendererInterface::SetSize was completely unused.

cricket::VideoRenderer::SetSize only had dummy implementations
returning true and doing nothing.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1574963002

Cr-Commit-Position: refs/heads/master@{#11298}
2016-01-19 08:52:55 +00:00
honghaiz
cec0a08275 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
Plus, in stunport, turnport and allocation sequence, create a socket using the new interface.

BUG=

Review URL: https://codereview.webrtc.org/1556743002

Cr-Commit-Position: refs/heads/master@{#11279}
2016-01-15 22:49:15 +00:00
glaznev
f4decb5469 Add QP statistics logging to Android HW encoder.
- Similar to HW decoder log timing for first 5 frames
of HW encoder input and output to track correct encoder
start.
- Increment input timestamp when frame is dropped.
- Better calculate average encoding time.
- Use different tags for encoder and decoder modules.

BUG=b/26504665

Review URL: https://codereview.webrtc.org/1587943004

Cr-Commit-Position: refs/heads/master@{#11277}
2016-01-15 21:49:26 +00:00
ivoc
d66b44d565 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}

Review URL: https://codereview.webrtc.org/1540103002

Cr-Commit-Position: refs/heads/master@{#11267}
2016-01-15 11:06:41 +00:00
Honghai Zhang
20ac434010 Fix a test bot failure.
That was caused by https://codereview.webrtc.org/1581903002/

BUG=
R=pthatcher@google.com
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1587373002 .

Cr-Commit-Position: refs/heads/master@{#11260}
2016-01-14 23:56:35 +00:00
honghaiz
67b1e1ab0b Put options as the argument of the java PeerConnectionFactory constructor.
BUG=

Review URL: https://codereview.webrtc.org/1581903002

Cr-Commit-Position: refs/heads/master@{#11257}
2016-01-14 22:45:44 +00:00
phoglund
37ebcf0ce5 Reland "Add APK targets to build libjingle tests for Android."
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/

This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391.

We have made more preparations downstream, so this should work now. Original CL by perkj@.

BUG=webrtc:2365
The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/

Review URL: https://codereview.webrtc.org/1570513004

Cr-Commit-Position: refs/heads/master@{#11186}
2016-01-08 13:05:01 +00:00
perkj
fbeb97e01f Fix clang warning in peerconnection_jni.cc
TEST= export GYP_DEFINES="OS=android clang=1" ...
      ninja -C out/Debug AppRTCDemo
BUG=webrtc:5399

Review URL: https://codereview.webrtc.org/1561073005

Cr-Commit-Position: refs/heads/master@{#11181}
2016-01-08 08:43:15 +00:00
Taylor Brandstetter
e5ba13bc09 Adding a way for a Java RtpSender to set a track without taking ownership.
This means that the track will still have a reference count after the
PeerConnection and RtpSender have been destroyed.

R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1566103003 .

Cr-Commit-Position: refs/heads/master@{#11176}
2016-01-07 23:11:33 +00:00
kjellander
60ca31bf5d Roll chromium_revision d66326c..4df108a (367167:367307)
The changes in d66326c..4df108a/build/common.gypi
enables a lot more warnings, which have been disabled/fixed in this CL.
See tracking bugs for remaining work.

Change log: d66326c..4df108a
Full diff: d66326c..4df108a

Changed dependencies:
* src/buildtools: fee7f1e..6d0c448
* src/third_party/libsrtp: b8dd754..8a7662a
DEPS diff: d66326c..4df108a/DEPS

No update to Clang.

BUG=webrtc:5397, webrtc:5398, webrtc:5399
TBR=hta@webrtc.org, perkj@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1553033002

Cr-Commit-Position: refs/heads/master@{#11147}
2016-01-04 18:16:01 +00:00
ivoc
a4df27b671 Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
Reason for revert:
Compile error on Android needs to be fixed before relanding.

Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}

TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1537213002

Cr-Commit-Position: refs/heads/master@{#11094}
2015-12-19 18:14:18 +00:00
ivoc
f4f5cb0927 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

NOTRY=true
TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1541633002

Cr-Commit-Position: refs/heads/master@{#11093}
2015-12-19 18:02:39 +00:00
deadbeef
bd7d8f7e2b Adding a MediaStream parameter to createSender.
This will allow an app to create senders with the same stream id,
without SDP munging.

Review URL: https://codereview.webrtc.org/1538673002

Cr-Commit-Position: refs/heads/master@{#11092}
2015-12-19 00:58:51 +00:00
ivoc
36d4c54500 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.

Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}

TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1533913004

Cr-Commit-Position: refs/heads/master@{#11087}
2015-12-18 16:05:21 +00:00
Peter Boström
b7d9a97ce4 Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.

BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1406903002 .

Cr-Commit-Position: refs/heads/master@{#11084}
2015-12-18 15:01:23 +00:00
ivoc
ae2c5ad12a Added option to specify a maximum file size when recording an AEC dump.
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.

BUG=webrtc:4741
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1413483003

Cr-Commit-Position: refs/heads/master@{#11081}
2015-12-18 11:53:42 +00:00
perkj
88518a22c6 Use NV21 instead of YUV12 and clean up.
BUG=webrtc:5375

Review URL: https://codereview.webrtc.org/1530843002

Cr-Commit-Position: refs/heads/master@{#11079}
2015-12-18 08:37:10 +00:00
perkj
48477c1c6a MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a texture.
BUG=webrtc:4993

Review URL: https://codereview.webrtc.org/1523843006

Cr-Commit-Position: refs/heads/master@{#11078}
2015-12-18 08:34:44 +00:00
kwiberg
0eb15ed7b8 Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!

This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.

Review URL: https://codereview.webrtc.org/1460043002

Cr-Commit-Position: refs/heads/master@{#11064}
2015-12-17 11:04:24 +00:00
perkj
672aba3f57 Fix error prone code in VideoCapturerAndroid
BUG=webrtc:5282

Review URL: https://codereview.webrtc.org/1486423003

Cr-Commit-Position: refs/heads/master@{#11046}
2015-12-16 10:17:24 +00:00
Magnus Jedvert
51254331cc Android: Refactor renderers to allow apps to inject custom shaders
This CL:
 * Abstracts the functions in GlRectDrawer to an interface.
 * Adds viewport location as argument to the draw() functions, because this information may be needed by some shaders. This also moves the responsibility of calling GLES20.glViewport() to the drawer.
 * Moves uploadYuvData() into a separate helper class.
 * Adds new SurfaceViewRenderer.init() function and new VideoRendererGui.create() function that takes a custom drawer as argument. Each YuvImageRenderer in VideoRendererGui now has their own drawer instead of a common one.

BUG=b/25694445
R=nisse@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1520243003 .

Cr-Commit-Position: refs/heads/master@{#11031}
2015-12-15 15:22:38 +00:00
perkj
9638143033 Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ )
Reason for revert:
Clients have been updated.

Original issue's description:
> Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ )
>
> Reason for revert:
> Revert due breaking other clients.
>
> Original issue's description:
> > Made EglBase an abstract class and cleaned up.
> > Adds EglBase10 that implemenents EglBase for EGL 1.0
> >
> > BUG=webrtc:4993
> > TBR=glaznew@webrtc.org
> >
> > Committed: https://crrev.com/3207916f35ded33f586774e2c98d4d0089fe3c6e
> > Cr-Commit-Position: refs/heads/master@{#11011}
>
> TBR=magjed@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4993
>
> Committed: https://crrev.com/e22e1cb399748112f308b488e7535754ef6b807d
> Cr-Commit-Position: refs/heads/master@{#11013}

TBR=magjed@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4993

Review URL: https://codereview.webrtc.org/1522303004

Cr-Commit-Position: refs/heads/master@{#11024}
2015-12-15 10:48:13 +00:00
perkj
e22e1cb399 Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ )
Reason for revert:
Revert due breaking other clients.

Original issue's description:
> Made EglBase an abstract class and cleaned up.
> Adds EglBase10 that implemenents EglBase for EGL 1.0
>
> BUG=webrtc:4993
> TBR=glaznew@webrtc.org
>
> Committed: https://crrev.com/3207916f35ded33f586774e2c98d4d0089fe3c6e
> Cr-Commit-Position: refs/heads/master@{#11011}

TBR=magjed@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4993

Review URL: https://codereview.webrtc.org/1522073002

Cr-Commit-Position: refs/heads/master@{#11013}
2015-12-14 14:43:39 +00:00
perkj
3207916f35 Made EglBase an abstract class and cleaned up.
Adds EglBase10 that implemenents EglBase for EGL 1.0

BUG=webrtc:4993
TBR=glaznew@webrtc.org

Review URL: https://codereview.webrtc.org/1526463002

Cr-Commit-Position: refs/heads/master@{#11011}
2015-12-14 14:21:19 +00:00
stefan
bc14164aad Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
Reason for revert:
Breaks bots.

Original issue's description:
> Add APK targets to build libjingle_peerconnection_unittests for Android.
>
> BUG=webrtc:2365
>
> The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/
>
> Committed: https://crrev.com/a78c0211fd50369a75a962385db6163bd8ded239
> Cr-Commit-Position: refs/heads/master@{#11007}

TBR=kjellander@webrtc.org,tommi@webrtc.org,perkj@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2365

Review URL: https://codereview.webrtc.org/1521993002

Cr-Commit-Position: refs/heads/master@{#11009}
2015-12-14 12:31:22 +00:00
perkj
a78c0211fd Add APK targets to build libjingle_peerconnection_unittests for Android.
BUG=webrtc:2365

The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/

Review URL: https://codereview.webrtc.org/1511633002

Cr-Commit-Position: refs/heads/master@{#11007}
2015-12-14 10:41:37 +00:00
Per
71f5a9a377 This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers.
Ie, rotation is applied in C++ in the VideoFrameFactory is  apply_rotation_ is set. If not, rotation is sent in RTP.

BUG=webrtc:4993
R=nisse@chromium.org

Review URL: https://codereview.webrtc.org/1493913007 .

Cr-Commit-Position: refs/heads/master@{#10986}
2015-12-11 08:32:50 +00:00
perkj
796cfaf7f7 Add VideoCodec::PreferDecodeLate
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.

Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.

Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.

Review URL: https://codereview.webrtc.org/1428293003

Cr-Commit-Position: refs/heads/master@{#10974}
2015-12-10 17:27:45 +00:00
nisse
c490e01bd1 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to
do the conversion using an opengl fragment shader.

BUG=webrtc:4993

Review URL: https://codereview.webrtc.org/1460703002

Cr-Commit-Position: refs/heads/master@{#10972}
2015-12-10 14:23:42 +00:00
perkj
7b2f7627e4 Don't call SetPreviewFormat if capturing to textures.
This fix an issue seen on Huawei Y300 where the camera feed is black and white if we capture to textures and setpreviewformat is called.

BUG=webrtc:4993

Review URL: https://codereview.webrtc.org/1502223002

Cr-Commit-Position: refs/heads/master@{#10941}
2015-12-08 20:03:07 +00:00
Peter Boström
6f28cf0b95 Implement standalone event tracing in AppRTCDemo.
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.

BUG=webrtc:5158
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1457383002 .

Cr-Commit-Position: refs/heads/master@{#10921}
2015-12-07 22:17:26 +00:00
nisse
03f80ebb83 Refactor EglBase configuration.
Delete EglBase.ConfigType, instead pass arrays of attributes, and define
constant arrays for the common cases.

Both in progress NativeToI420 and extending GlRectDrawer to other shapes (with alpha) needs this.

BUG=b/25694445

Review URL: https://codereview.webrtc.org/1498003002

Cr-Commit-Position: refs/heads/master@{#10908}
2015-12-07 09:17:22 +00:00