Commit Graph

7971 Commits

Author SHA1 Message Date
tommi@webrtc.org
66f153f89f Make LS_ logging constants to match Chromium's logging constants when building with Chrome.
This was causing logging to be done at incorrect levels and filters not work as expected.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40239004

Cr-Commit-Position: refs/heads/master@{#8635}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8635 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 15:56:46 +00:00
pbos@webrtc.org
a2a6fe66a3 Reconfigure default streams on AddRecvStream.
Makes sure RTX can be used for streams that have received early media
before being properly configured.

BUG=1788
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46499004

Cr-Commit-Position: refs/heads/master@{#8634}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8634 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 15:35:48 +00:00
perkj@webrtc.org
bcead305a2 Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.

Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.

With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame

BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429004

Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 12:38:22 +00:00
kjellander@webrtc.org
75e850e192 Enable isac NEON building on Aarch64
Passed building isac_neon and modules_unittests on Android ARM64 and ARMv7.
Passed modules_unittests with following filters:
  --gtest_filter=FiltersTest*
  --gtest_filter=LpcMaskingModelTest*
  --gtest_filter=TransformTest*
  --gtest_filter=FilterBanksTest*

WebRtcIsacfix_CalculateResidualEnergyNeon is not enabled due to Issue 4224.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39979004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#8632}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8632 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 12:29:23 +00:00
tommi@webrtc.org
0d5ea21325 Remove lock from Bitrate() and FrameRate() in VideoSender.
These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder.  It's therefore safe to not require a lock to access _encoder on this thread.

I'm making access to the rate variables from VCMGenericEncoder, thread safe, by using a lock that's not associated with the encoder.  There should be little to no contention there.  While modifying VCMGenericEncoder, I noticed that a couple of member variables weren't needed, so I removed them.

The reason for this change is that getStats is currently contending with the encoder when Bitrate() is called. On my machine, this means that getStats can take about 25-30ms instead of ~1ms.

Also adding some documentation for other methods and a suggestion for how we could avoid contention between the encoder and the network thread.

BUG=2822
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43479004

Cr-Commit-Position: refs/heads/master@{#8631}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8631 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 12:21:41 +00:00
magjed@webrtc.org
f98030b029 Add intermediate TextureVideoFrame typedef for Chromium
BUG=1128
R=perkj@webrtc.org
TBR=stefan

Review URL: https://webrtc-codereview.appspot.com/42239004

Cr-Commit-Position: refs/heads/master@{#8630}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8630 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 11:18:43 +00:00
magjed@webrtc.org
45cdcce5f5 Remove TextureVideoFrame
TextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of TextureVideoFrame into the base class. Then it's possible to completely remove TextureVideoFrame and all its files. Also, there is no point in having I420VideoFrame virtual anymore.

R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/40229004

Cr-Commit-Position: refs/heads/master@{#8629}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8629 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 10:41:47 +00:00
kjellander@webrtc.org
7158ec1727 Remove android-webrtc.mk
This is a stale file that haven't been used for years.
If anyone relies on this for their custom build, it's better
they keep it in their repo instead.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45509004

Cr-Commit-Position: refs/heads/master@{#8628}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8628 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 08:44:44 +00:00
kjellander@webrtc.org
e41ec818a7 Remove libjingle_root GYP variable
It is no longer needed.

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

Review URL: https://webrtc-codereview.appspot.com/44449004

Cr-Commit-Position: refs/heads/master@{#8627}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8627 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 08:03:47 +00:00
henrik.lundin@webrtc.org
e9217b4bdb Remove WebRtcACMEncodingType
The parameter was not needed; it was sufficient with a bool indicating
speech or not speech. This change propagates to the InFrameType
callback function. Some tests are updated too.

COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42209004

Cr-Commit-Position: refs/heads/master@{#8626}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8626 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 07:51:21 +00:00
kjellander@webrtc.org
84f5309dbd Roll chromium_revision e144d30..e8ef1d1 (318658:319252)
Relevant changes:
* src/buildtools: 93b3d0a..d4dd4f7
* src/third_party/icu: df1bf38..eda9e75
* src/tools/grit: a5890a8..0287c18
* src/tools/gyp: 3464008..4a9b712
Details: e144d30..e8ef1d1/DEPS

Clang version changed 223108:231191
Details: e144d30..e8ef1d1/tools/clang/scripts/update.sh

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43589004

Cr-Commit-Position: refs/heads/master@{#8625}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8625 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 07:18:14 +00:00
kjellander@webrtc.org
a743f6f17f Widening memcheck suppressions for libjingle_peerconnection_unittest
Make the excluded tests filter match the ones for Dr Memory, since
these tests are probably flaky only due to the same slowdown caused
by memcheck as for Dr Memory.

BUG=4387
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46469004

Cr-Commit-Position: refs/heads/master@{#8624}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8624 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 07:16:46 +00:00
pthatcher@webrtc.org
818c4984e4 Modify the simulcast encoder factory adapter to allow external encoder factories that support more than one codec.
Only VP8 encoders will be wrapped in the simulcast adapter; other codec types will be created directly with the real encoder factory and cleaned up appropriately.

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

Review URL: https://webrtc-codereview.appspot.com/40169004

Cr-Commit-Position: refs/heads/master@{#8623}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8623 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 02:21:23 +00:00
marpan@webrtc.org
16a87b97f9 Add VP9 denoiser test to videoprocessor_integrationtest.
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43599004

Cr-Commit-Position: refs/heads/master@{#8622}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8622 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 22:19:15 +00:00
aluebs@webrtc.org
1d88394bcb Add support for arbitrary array geometries in Beamformer
R=andrew@webrtc.org, mgraczyk@chromium.org

Review URL: https://webrtc-codereview.appspot.com/38299004

Cr-Commit-Position: refs/heads/master@{#8621}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8621 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 20:39:20 +00:00
andrew@webrtc.org
0933d01d09 Enabling common_audio building with NEON on ARM64
Passed building common_audio_neon and common_audio_unittests both on
Android ARMv7 and Android ARM64. Pass common_audio_unittests tests both
on Android ARMv7 and Android ARM64.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Change-Id: I8e0722f356db8cca6fc8232f00ae1e898a086f5a

Review URL: https://webrtc-codereview.appspot.com/40629004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#8620}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8620 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 19:14:21 +00:00
bjornv@webrtc.org
d7a212e8b9 audio_processing/aec: Increased delay metrics aggregation window to five seconds
The known clients (GetStats and UMA histogram in Chrome) use at least 5 second aggregation window. There is no particular value in calculating the metrics more often.

The CL also includes a small refactoring moving a declaration inside an if statement.

BUG=2994
TEST=N/A
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40219004

Cr-Commit-Position: refs/heads/master@{#8619}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8619 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 16:14:58 +00:00
stefan@webrtc.org
c3f15c08bc Fix scoped_ptrs in bwe_simulations.
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45469004

Cr-Commit-Position: refs/heads/master@{#8618}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8618 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 16:06:21 +00:00
kjellander@webrtc.org
74304330df Print better information during Chromium sync.
Many users are having problems with the initial sync.
Let's print a bit more information and advice on how
to recover from an aborted sync.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42229004

Cr-Commit-Position: refs/heads/master@{#8617}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8617 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 14:38:42 +00:00
magjed@webrtc.org
2386d6dd92 Revert 8599 "Revert 8580 "Unify underlying frame buffer in I420VideoFrame and...""
It's possible to build Chrome on Windows with this patch now.

BUG=1128

> 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

R=pbos@webrtc.org
TBR=mflodman, pbos, perkj, tommi

Review URL: https://webrtc-codereview.appspot.com/45489004

Cr-Commit-Position: refs/heads/master@{#8616}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8616 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 14:03:51 +00:00
pbos@webrtc.org
67a9e40286 Prevent encoding frames with wrong resolution.
This is a speculative fix for a crash that should be able to happen if a
codec is reconfigured while a frame is leaving the
VideoProcessingModule, causing a mismatch between configured codec and
input frame size.

BUG=
R=magjed@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48379004

Cr-Commit-Position: refs/heads/master@{#8615}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8615 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 13:58:16 +00:00
tommi@webrtc.org
03054486f5 Adding basic support for posting tasks to a process thread.
BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41099004

Cr-Commit-Position: refs/heads/master@{#8614}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8614 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 13:14:19 +00:00
tommi@webrtc.org
658d2015f3 Allow VideoSender to be constructed on one thread but initialized and used for doing registrations, on another.
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42219004

Cr-Commit-Position: refs/heads/master@{#8613}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8613 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 12:22:22 +00:00
kjellander@webrtc.org
7008f2227c Revert Clang roll in r8596 + add memcheck suppression.
The roll seems to cause leaks on our Linux Memcheck bot.
Added a suppression needed for Trusty in order to run
memcheck similar to the bot (that runs Precise).
Leave all the other source code edits from r8596 in place.

See also http://chromegw/i/client.webrtc/builders/Linux%20Memcheck/builds/3343

TBR=pbos@webrtc.org
TESTED=Can no longer repro memcheck failure with this patch applied:
GYP_DEFINES="build_for_tool=memcheck" webrtc/build/gyp_webrtc
ninja -C out/Release libjingle_peerconnection_unittest
tools/valgrind-webrtc/webrtc_tests.sh --test libjingle_peerconnection_unittest --tool memcheck --target Release --build-dir out  --gtest_filter=WebRtcSessionTest.TestIncorrectMLinesInLocalAnswer

Review URL: https://webrtc-codereview.appspot.com/47419004

Cr-Commit-Position: refs/heads/master@{#8612}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8612 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 08:53:44 +00:00
tommi@webrtc.org
5af41aabae Fix uninitialized variable. If FindConstraint() returns false, we check |value| in two places and at that point, it can hold an uninitialized value. Caught by Linux Memcheck builder.
http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/3351/steps/libjingle_peerconnection_unittest/logs/0A34BA777AB03D08

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43579004

Cr-Commit-Position: refs/heads/master@{#8611}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8611 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 08:42:13 +00:00
andrew@webrtc.org
fa67463d37 skip isac_neon if neon is not supported
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39909004

Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

Cr-Commit-Position: refs/heads/master@{#8610}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8610 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 06:07:51 +00:00
guoweis@webrtc.org
bbce5efaa6 Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8582

Committed: https://code.google.com/p/webrtc/source/detail?r=8607

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

Cr-Commit-Position: refs/heads/master@{#8609}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8609 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 04:39:07 +00:00
guoweis@webrtc.org
d43b2c098d Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit 86c33e3a94.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/47409004

Cr-Commit-Position: refs/heads/master@{#8608}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8608 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 04:03:44 +00:00
guoweis@webrtc.org
86c33e3a94 Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8582

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

Cr-Commit-Position: refs/heads/master@{#8607}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8607 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 03:40:48 +00:00
guoweis@webrtc.org
4536289353 Add CVO support to RTP sender side.
According to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf,
CVO byte should only be added in the last packet of each key frame or when the rotation changes. Currently, we're adding this byte in each frame to start with.

BUG=4145
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42439004

Cr-Commit-Position: refs/heads/master@{#8606}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8606 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 22:55:43 +00:00
jiayl@webrtc.org
61e00b0bca 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

Committed: https://code.google.com/p/webrtc/source/detail?r=8581

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

Cr-Commit-Position: refs/heads/master@{#8605}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8605 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 22:18:18 +00:00
marpan@webrtc.org
6daacbc8ae Set cpu_speed parameter for low resolutions, for non-simulcast.
Allow for setting different cpu_speed setting based on resolution, for non-simulcast.
Use the existing low resolution simulcast cpu_speed setting for the non-simulcast case.

No change to simulcast behavior, unless top/highest layer stream is also below CIF resolution,
(in which case all layers will use lower the cpu_speed setting =-4).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37319004

Cr-Commit-Position: refs/heads/master@{#8603}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8603 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 21:47:18 +00:00
kjellander@webrtc.org
7b93ea1667 Remove DCHECK from common_types.cc
The DCHECKs added in https://webrtc-codereview.appspot.com/41069004/
introduces a dependency on base, which may cause problems.
This CL reverts that change.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48389004

Cr-Commit-Position: refs/heads/master@{#8602}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8602 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 20:10:38 +00:00
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 bbbdeed2bf.

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