7988 Commits

Author SHA1 Message Date
tommi@webrtc.org
4a4e688438 Remove dependecy on win32.h in criticalsection.h. This was causing build errors that we haven't fully figured out yet but somehow this caused override files to include the files they're supposed to override, which in turn included webrtc build files that then conflict with Chromium's configuration.
TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45479004

Cr-Commit-Position: refs/heads/master@{#8601}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8601 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 20:10:37 +00:00
tommi@webrtc.org
f7abb12aa9 Fix OVERRIDE->override again after reverting video frame cl.
TBR=magjed@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40199004

Cr-Commit-Position: refs/heads/master@{#8600}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8600 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 17:43:49 +00:00
tommi@webrtc.org
1f94407319 Revert 8580 "Unify underlying frame buffer in I420VideoFrame and..."
This is unfortunately causing build problems in Chrome on Windows.

> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
> 
> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
> 
> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
> 
> Some additional minor changes are:
> * Disallow creation of 0x0 texture frames.
> * Remove the half-implemented ref count functions in I420VideoFrame.
> * Remove the Alias functionality in WebRtcVideoFrame
> 
> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
> 
> BUG=1128
> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/42469004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42199005

Cr-Commit-Position: refs/heads/master@{#8599}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8599 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 17:35:00 +00:00
henrik.lundin@webrtc.org
c86bbbaa93 Add speech flag to EncodedInfo
The flag indicates if the encoded bitstream is speech or comfort noise.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42629004

Cr-Commit-Position: refs/heads/master@{#8598}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8598 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 16:03:19 +00:00
tommi@webrtc.org
92f4018d80 Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
Rename "AddValue" methods to AddXxx where Xxx is the type being added. Moving forward, we'll support those types natively without conversion to string.

Normalizing the extraction code to have fewer places that add the same stats and data driven additions to reports instead of multiple call sites.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47369004

Cr-Commit-Position: refs/heads/master@{#8597}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8597 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 15:25:44 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
stefan@webrtc.org
792f1a14e2 Break out allocation from BitrateController into a BitrateAllocator.
This also refactors some of the padding and allocation code in ViEEncoder, and
makes ChannelGroup a simple forwarder from BitrateController to
BitrateAllocator.

This CL is part of a bigger picture, see https://review.webrtc.org/35319004/ for
details.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44399004

Cr-Commit-Position: refs/heads/master@{#8595}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8595 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 12:25:17 +00:00
henrik.lundin@webrtc.org
61c22aca5f Eliminate AcmGenericCodec::Add10MsData
All encoding work is now done in the Encode function.

Note: This CL leaves a technical debt in
AudioCodingModuleImpl::Add10MsData. This will be fixed in later
changes.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46419004

Cr-Commit-Position: refs/heads/master@{#8594}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8594 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 11:52:17 +00:00
magjed@webrtc.org
f82109cb4d Initialize memory in I420VideoFrame unittest
Previously, when CreateEmptyFrame was called with a smaller size than before, we would reuse the allocation. Now, we allocate a new tight frame. The CL that made this change is https://webrtc-codereview.appspot.com/42469004/. This exposed an uninitialized memory problem in a I420VideoFrame unittest. This CL fixes that unittest.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42619004

Cr-Commit-Position: refs/heads/master@{#8593}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8593 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 11:30:48 +00:00
magjed@webrtc.org
487afc704d Always define RTC_NOTREACHED, not just in non-chromium builds
R=tommi@webrtc.org
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/40189004

Cr-Commit-Position: refs/heads/master@{#8592}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8592 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 11:26:19 +00:00
magjed@webrtc.org
9cd7c26d1a Rename NOTREACHED to RTC_NOTREACHED to avoid name conflict with Chromium
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45459004

Cr-Commit-Position: refs/heads/master@{#8591}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8591 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 10:21:19 +00:00
kjellander@webrtc.org
6dab6d700d Let Chromium declare the mips_dsp_rev build variable.
In https://codereview.chromium.org/883253003, the mips_dsp_rev
build variable is added to Chromium's GYP and GN build files. Remove
the declarations of mips_dsp_rev from WebRTC's GYP and GN build files.

Replace mips_fpu with mips_float_abi and remove the compiler flags that
are already set by Chromium.

The main review of this was done in https://webrtc-codereview.appspot.com/39779004
but since that CL wasn't created with the right base URL, I made
this in order to be able to run WebRTC trybots properly.

BUG=446234
TBR=wtc@chromium.org

Review URL: https://webrtc-codereview.appspot.com/44549004

Cr-Commit-Position: refs/heads/master@{#8590}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8590 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 09:51:17 +00:00
henrik.lundin@webrtc.org
1d25c87199 Reland r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
This effectively reverts r8578.

TBR=jmarusic@webrtc.org

Original commit message:
Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac

With this change, support for iSAC-RED is incorporated into the
regular AudioEncoderDecoderIsac class.

COAUTHOR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44539004

Cr-Commit-Position: refs/heads/master@{#8589}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8589 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:42 +00:00
pbos@webrtc.org
058b1f17ac Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.

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

Review URL: https://webrtc-codereview.appspot.com/47379004

Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:16 +00:00
kjellander@webrtc.org
7572d85aa8 rtc_unittests on Android
BUG=4364
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42549004

Cr-Commit-Position: refs/heads/master@{#8587}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8587 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:48:17 +00:00
kjellander@webrtc.org
c98f6f368a PRESUBMIT: Exclude overrides paths from source above GYP check.
Since base.gyp contains a couple of those.

BUG=4185
TESTED=Ran git cl presubmit with a modified base.gyp without this patch - got errors about overrides. Ran another time with this patch applied - no errors.
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44529004

Cr-Commit-Position: refs/heads/master@{#8586}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8586 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 07:08:44 +00:00
guoweis@webrtc.org
fc2f146af2 Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit bbbdeed2bff31777ca7d298d17336fe94626f5b3.

TBR=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41109004

Cr-Commit-Position: refs/heads/master@{#8585}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8585 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 04:50:41 +00:00
pthatcher@webrtc.org
7bea1ffe77 Expose negotiated ciphers through stats API.
Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.

BUG=3976
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35169004

Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 01:38:49 +00:00
jiayl@webrtc.org
be77872d2c Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
Breaking Chromium FYI.

TBR=pthatcher@webrtc.org

This reverts commit 369f68255ffd3d6f3e449e0defeae820cefd4f29.

BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004


Review URL: https://webrtc-codereview.appspot.com/47389004

Cr-Commit-Position: refs/heads/master@{#8583}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8583 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 00:19:16 +00:00
guoweis@webrtc.org
bbbdeed2bf Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8582}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8582 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 23:27:41 +00:00
jiayl@webrtc.org
369f68255f Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8581}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8581 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 23:14:18 +00:00
magjed@webrtc.org
c8895aa2f3 Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.

This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.

Some additional minor changes are:
* Disallow creation of 0x0 texture frames.
* Remove the half-implemented ref count functions in I420VideoFrame.
* Remove the Alias functionality in WebRtcVideoFrame

The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
* Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
* Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.

BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42469004

Cr-Commit-Position: refs/heads/master@{#8580}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8580 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 21:22:26 +00:00
jiayl@webrtc.org
8ad96605c1 Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
Test failure: http://chromegw/i/client.webrtc/builders/Linux32%20Release/builds/3557

This reverts commit df512cc8b73ff519dcdf63a2603ab312d3443402.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41089004

Cr-Commit-Position: refs/heads/master@{#8579}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8579 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 20:35:34 +00:00
henrik.lundin@webrtc.org
bcef431902 Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
Some of the build bots seems to have reacted to this change.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42169004

Cr-Commit-Position: refs/heads/master@{#8578}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8578 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 20:13:48 +00:00
henrik.lundin@webrtc.org
1fc28f2305 Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
With this change, support for iSAC-RED is incorporated into the regular
AudioEncoderDecoderIsac class.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43549004

Cr-Commit-Position: refs/heads/master@{#8577}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8577 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 19:31:17 +00:00
jiayl@webrtc.org
df512cc8b7 Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8576}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8576 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 16:42:21 +00:00
pbos@webrtc.org
982cd2a94c Filter receiver-side DataCountersUpdated on SSRC.
BUG=1788,1667
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44509004

Cr-Commit-Position: refs/heads/master@{#8575}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8575 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 15:57:18 +00:00
sprang@webrtc.org
b144b4b74e Fixed bug in SendTimeHistory, where deleting packets via the getter
would not update the oldest suence number.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42589004

Cr-Commit-Position: refs/heads/master@{#8574}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8574 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 15:44:54 +00:00
minyue@webrtc.org
0561716ae2 Adding Opus DTX support in ACM.
This solution does not use the existing VAD/DTX logic of ACM, since Opus DTX is codec feature, while ACM VAD/DTX is mainly for setting the WebRTC VAD/DTX.

During the development of this CL, two old bugs were found and are fixed in this CL too.

They are in
webrtc/modules/audio_coding/test/Channels.cc
and webrtc/modules/audio_coding/main/acm2/acm_opus_unittest.cc
respectively.

BUG=webrtc:1014
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38469004

Cr-Commit-Position: refs/heads/master@{#8573}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8573 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 12:03:14 +00:00
perkj@webrtc.org
a1c9803e32 Fix crash in setPictureSize on Galaxy Nexus.
This cl tries to find the best supported pictureSize before setting it.
BUG=4197
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45419004

Cr-Commit-Position: refs/heads/master@{#8571}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8571 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 10:54:21 +00:00
perkj@webrtc.org
be00e3c198 Make sure VideoFrameFactory handles rotated frames when scaling.
BUG=4366
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41079004

Cr-Commit-Position: refs/heads/master@{#8570}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8570 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 10:48:46 +00:00
kwiberg@webrtc.org
9e5f941ff1 Remove webrtc/system_wrappers/interface/scoped_ptr.h
No one uses it anymore; they've all switched to the cooler
webrtc/base/scoped_ptr.h instead.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42129004

Cr-Commit-Position: refs/heads/master@{#8569}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8569 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 10:17:49 +00:00
perkj@webrtc.org
1f914ecf9a Remove suppression for WebRtcVideoFrameTest::TestInit
The problem is fixed in https://webrtc-codereview.appspot.com/41029004/.

bug: https://code.google.com/p/libyuv/issues/detail?id=377

R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46389004

Cr-Commit-Position: refs/heads/master@{#8568}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8568 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 09:39:44 +00:00
minyue@webrtc.org
db93b68031 Removing NetEq's direct dependencies on Opus headers.
Neteq had a direct dependency on Chromium/third_party/opus. This should be relayed by target webrtc_opus.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42529004

Cr-Commit-Position: refs/heads/master@{#8567}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8567 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 09:28:53 +00:00
magjed@webrtc.org
cb04aa4a81 WebRtcVideoFrameTest: Initialize memory to fix DrMemory error
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41029004

Cr-Commit-Position: refs/heads/master@{#8566}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8566 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 09:24:16 +00:00
kjellander@webrtc.org
909f494b43 Roll chromium_revision 2c3ffb2..e144d30 (317530:318658)
Relevant changes:
* src/third_party/boringssl/src: b180ee9..09bdb2a
* src/third_party/icu: 2081ee6..df1bf38
* src/third_party/libvpx: 33bbffe..080710f
* src/third_party/openmax_dl: 21c8abe..6658243
Details: 2c3ffb2..e144d30/DEPS

Clang version was not updated in this roll.

BUG=webm:962
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43519004

Cr-Commit-Position: refs/heads/master@{#8565}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8565 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 07:21:14 +00:00
perkj@webrtc.org
1d82813961 Reland "Fix CVO in androidvideocapturer".
This cl was originally revieved in https://webrtc-codereview.appspot.com/40759004/

Patchset 2 adds a unittest for VideoFrame::Reset with and without the apply_rotation flag set.

BUG=4145
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42559004

Cr-Commit-Position: refs/heads/master@{#8564}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8564 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 06:44:40 +00:00
aluebs@webrtc.org
c9ce07ed87 Add Config option to enable 48kHz support in AudioProcessing
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45389004

Cr-Commit-Position: refs/heads/master@{#8563}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8563 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 20:07:51 +00:00
tommi@webrtc.org
0482d01902 Implement TraceCallback in a nested class of WebRtcVideoEngine.
This is to fix a race that occurs in unit tests when the tests inherit
from the engine class that also implements the callback interface for
tracing.  If tracing happens while the most derived class is still being
constructed, we're in trouble.

So, instead, factoring out the TraceCallback implementation.

R=pbos@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43489004

Cr-Commit-Position: refs/heads/master@{#8562}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8562 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 17:51:44 +00:00
magjed@webrtc.org
97ed2a4b70 I420VideoFrame: Remove function ResetSize
This is a partial reland of https://webrtc-codereview.appspot.com/39939004/.

The original CL was reverted because ViECapturer use ResetSize/IsZeroSize on |captured_frame_| as a check to make sure each captured frame is only delivered once. Removing ResetSize introduced a race condition where a captured frame could be delivered multiple times.

I have fixed this problem in this CL by replacing ResetSize with scoped_ptr::release.

BUG=4352
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39359004

Cr-Commit-Position: refs/heads/master@{#8561}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8561 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 17:33:41 +00:00
glaznev@webrtc.org
43f4a47c28 Add more Android peer connection client unit tests:
- Add front/back camera switch test.
- Add video source stop and restart test to simulate
application going into background.
- Add a loopback test for 3 video codecs - VP8, VP8, H.264.
- Add a loopback test for voice only call.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43459004

Cr-Commit-Position: refs/heads/master@{#8560}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8560 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 17:32:18 +00:00
bjornv@webrtc.org
976c0f3043 audio_processing/aec: NEON code should not be invoked if it is detectable, but is not NEON
There exist devices with runtime checks for NEON, but where the device is not NEON. One such device is Tegra2 on which currently NEON code is running.

This fix adds a missing feature check when initializing the AEC.

BUG=4304
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42159004

Cr-Commit-Position: refs/heads/master@{#8559}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8559 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 16:25:51 +00:00
stefan@webrtc.org
48ac226b9a Add support for writing h264 decoder input to file and parsing interleaved length/packet RTP dumps.
This is useful for debugging h264 input when we don't have an h264 decoder, as the resulting file should be possible to play back using mplayer. It is also often convenient to dump rtp packets in an interleaved format where the size of a packet is inserted before the actual payload.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42139004

Cr-Commit-Position: refs/heads/master@{#8558}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8558 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 16:19:15 +00:00
marpan@webrtc.org
3fe17d1598 Adjust a few thresholds for VP9 tests.
Needed for the upcoming libvpx roll.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/44479004

Cr-Commit-Position: refs/heads/master@{#8557}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8557 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 15:34:19 +00:00
magjed@webrtc.org
fd33293d58 I420VideoFrame: Remove functions set_width and set_height
This is a partial reland of https://webrtc-codereview.appspot.com/39939004/.

The functions set_width and set_height in I420VideoFrame are not needed and just add complexity.

R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41009004

Cr-Commit-Position: refs/heads/master@{#8556}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8556 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:57:44 +00:00
pbos@webrtc.org
f1f0d9a4cd Remove WebRtcVideoEngine::SetVoiceEngine.
Instead enforcing that a voice engine is set on construction. Apart from
simplifying the class this permits tracing to be set up in the
constructor without worrying about racing sets from SetVoiceEngine
later.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44489004

Cr-Commit-Position: refs/heads/master@{#8555}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8555 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:30:46 +00:00
andresp@webrtc.org
e8f50df6b9 Remove avi recorder and corresponding enable_video flags.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:07:44 +00:00
henrik.lundin@webrtc.org
f56c162310 Remove AudioCodingModule::Process()
An earlier change moved the encoding work from Process to
Add10MsData; process was just a no-op.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=henrika@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43439004

Cr-Commit-Position: refs/heads/master@{#8553}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8553 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 12:30:19 +00:00
sprang@webrtc.org
25dd1dbb9f Fixed bug in test frame generator, causing incorrect reuse of frame
object, in turn causing performance regression.

Plus a small optimization.

BUG=chromium:460954, 4329
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42499004

Cr-Commit-Position: refs/heads/master@{#8552}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8552 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 11:56:16 +00:00
perkj@webrtc.org
60f9d6f959 Revert "Add default implementation to VideoSourceInterface."
Chrome test mock has been updated so VideoSourceInterface can now be pure virtual again. This reverts commit ed8d52378c43a7a93e0d2ca586486ca06db9eabe.

R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45399004

Cr-Commit-Position: refs/heads/master@{#8551}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8551 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 11:33:41 +00:00