Commit Graph

6987 Commits

Author SHA1 Message Date
henrika@webrtc.org
2176db343c AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land)
This CL was incorrectly reverted in r7647 by the libjingle sync bot.

TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7717 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-18 13:22:28 +00:00
kjellander@webrtc.org
c56814fb2d Roll chromium_revision 91f1781..d8c9041
Relevant changes:
* buildtools: c27f95b..6ea835d
* third_party/icu: d8b2a9d..6242e2f
* tools/gyp: 487c0b6..b13d8f2
* tools/swarming_client: 1f8ba35..5b827c9
Details: 91f1781..d8c9041/DEPS

Clang version was not updated in this roll, although the
-Wunused-local-typedef warning was enabled by default.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7716 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-18 10:25:04 +00:00
aluebs@webrtc.org
087da13fe8 Add empty 3 band splitting filter API
This is only an empty API that will never be used. For now is 48kHz not supported in AudioProcessing. For that it needs to be added in InitializeLocked. But before the 3 band filter bank needs to be populated.

BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 23:01:23 +00:00
pkasting@chromium.org
2656bf813f Fix ExpectedQueueTimeMs() to avoid truncation or overflow.
BUG=none
TEST=none
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 22:21:14 +00:00
guoweis@webrtc.org
930e004a81 Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7693

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 19:42:14 +00:00
magjed@webrtc.org
c72a22c23d Add preliminary empty file videoframefactory.cc
The purpose of this CL is to add a new file in libjingle without breaking Chromium in the process. The plan is to do the following:
1. Land a no-op videoframefactory.cc in webrtc (this file).
2. Wait for it to roll into Chromium.
3. Modify libjingle.gyp in Chromium to include this file.
4. Make the real change in webrtc with the real implementation of this file.
5. Wait for the change to roll into Chromium.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7712 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 16:34:00 +00:00
pbos@webrtc.org
f5b56fbc41 Annotate COMPILE_ASSERT with __attribute__((unused)).
Also renames UNUSED -> ATTRIBUTE_UNUSED to be able to use this when
building peerconnection_jni.cc which apparently has this defined to
something else.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=4018

Review URL: https://webrtc-codereview.appspot.com/28039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 13:47:38 +00:00
minyue@webrtc.org
4ef22d1d29 Setting Opus FEC as default
BUG=3986
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7710 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 09:26:39 +00:00
henrik.lundin@webrtc.org
966a708b93 Use RtpFileSource in NetEqDecodingTest
This CL removes the dependency on the old NETEQTEST_RTPpacket class
from the NetEqDecodingTest code, and also removes the dependency
from the modules_unittests target to neteq_test_tools.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 09:08:38 +00:00
tommi@webrtc.org
4ec19e306a Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..."
This didn't compile on the FYI bots.  Example error:

FAILED: E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-with-manifests environment.x86 True chrome_child.dll "E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-wrapper environment.x86 False link.exe /nologo /IMPLIB:chrome_child.dll.lib /DLL /OUT:chrome_child.dll @chrome_child.dll.rsp" 2 mt.exe rc.exe "obj\chrome\chrome_child_dll.chrome_child.dll.intermediate.manifest" obj\chrome\chrome_child_dll.chrome_child.dll.generated.manifest
content_renderer.lib(content_renderer.webrtc_video_capturer_adapter.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.peerconnectionfactory.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.videocapturer.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.dummydevicemanager.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

chrome_child.dll : fatal error LNK1120: 1 unresolved externals


> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
> 
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
> 
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
> 
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
> 
> Committed: https://code.google.com/p/webrtc/source/detail?r=7702
> 
> Review URL: https://webrtc-codereview.appspot.com/29949004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-16 22:58:11 +00:00
magjed@webrtc.org
858dbbced2 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-16 18:21:51 +00:00
henrike@webrtc.org
6a782c2a46 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
TBR=guoweis@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 22:33:13 +00:00
aluebs@webrtc.org
be05c74ec8 Wrap the splitting filter in its own class
This doesn't change the behavior at all.
The logic behind this is having one class which manages all the splitting filters, because in the future we plan to add a 3 band one for 48kHz support.
It also breaks the dependency of the AudioBuffer with the filter states of these filters (which are going to be different for the 3 band one). The AudioBuffer is complicated enough and is going to need changes to support 3 bands in the future, so any simplification is a good idea.
On top of that it eliminates repeated code in the APM (now only iterating over channels, but then also deciding in how many bands to split). This should be managed by the AudioBuffer directly.

BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7705 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 22:18:10 +00:00
pbos@webrtc.org
67c22478a4 Disable EndToEnd.GetStats test.
Looks like this test exposes a bug in jitter buffer after enabling
multiple streams. Will disable to be able to debug it in peace and not
have to revert.

TBR=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/31009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7704 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 17:42:51 +00:00
magjed@webrtc.org
a73d746562 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..."
Rease for revert: failed internal test cases

> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
> 
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
> 
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
> 
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/29949004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 13:25:25 +00:00
magjed@webrtc.org
bbd8cad21f cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7702 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 12:10:46 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats.
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 11:52:04 +00:00
magjed@webrtc.org
35c1ace185 Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..."
Reason for revert is failed testcases:
WebRtcVideoEngineExtendedTestFake.ResetSimulcastSendCodecOnNewFrameSize
WebRtcVideoEngineExtendedTestFake.MultipleSendStreamsDifferentFormats

> WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
> 
> BUG=3936
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/30039004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7700 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 16:21:49 +00:00
kjellander@webrtc.org
a1f5b96351 Remove unnecessary copying of libjingle resource files.
This copying has probably not been needed since
https://code.google.com/p/webrtc/source/detail?r=7088

BUG=398
TESTED=Removed the top-level talk directory and ran
libjingle_media_unittest from the following working directories:
* checkout-root/src/out/Debug
* checkout-root/src
* checkout-root

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7699 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 15:53:08 +00:00
magjed@webrtc.org
52da44b7e6 WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
BUG=3936
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7698 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 15:43:11 +00:00
pbos@webrtc.org
49ff40e32e Make SetREMBData accept vector of SSRCs.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 14:42:37 +00:00
pbos@webrtc.org
a9c2d454bd Fix and enable CanReceiveFec test.
Test relied on the first protected media packet that was dropped to
actually be rendered, while rendering it could have been skipped on slow
systems due to newer frames being decoded before rendering happens.

R=stefan@webrtc.org
BUG=3269

Review URL: https://webrtc-codereview.appspot.com/25159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7696 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 14:40:15 +00:00
bjornv@webrtc.org
ee30082af8 Set correct sample rate in far_frame in audioproc tool.
One debug recording with non matching sample rates between render and capture revealed a bug in modules/audio_processing/test/process_test.cc
The far_frame (render audio frame) used was loaded with the capture rate instead of the render rate with a data length mismatch error as result.

BUG=N/A
TESTED=manually on linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7695 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 11:00:10 +00:00
kjellander@webrtc.org
52bb521b47 Update isolate files for Android APK tests.
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).

A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.

BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 08:35:05 +00:00
guoweis@webrtc.org
312614a438 Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 03:38:05 +00:00
jiayl@webrtc.org
90b9b08332 Fix a platform check to use WEBRTC_WIN instead of OS_WIN.
BUG=4006
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/25169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 20:53:00 +00:00
jiayl@webrtc.org
6ca6190be2 Fix a SCTP message reordering issue in datachannel.cc.
Previously DataChannel::SendQueuedDataMessages continues the loop of sending queued messages if the channel is blocked, which will cause message reordering if the channel becomes unblocked during the loop, i.e. messages attempted after the unblocking will be sent earlier than the older messages attempted before the unblocking.

BUG=3979
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7690 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 17:28:40 +00:00
magjed@webrtc.org
ea73ff7267 webrtc::Scaler: Preserve aspect ratio
BUG=3936
R=glaznev@webrtc.org, stefan@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7679

Review URL: https://webrtc-codereview.appspot.com/28969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7689 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 09:52:03 +00:00
magjed@webrtc.org
0b3d89b500 VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7688 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 08:58:49 +00:00
kjellander@webrtc.org
14ea50a8e3 Change the static_library("webrtc") to a source set in the GN build.
Static libraries cannot have only headers as sources (libtool complains
that there's nothing to actually link).

TBR=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7687 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 07:56:21 +00:00
andrew@webrtc.org
0e37b898f0 replace inline assembly WebRtcAecm_CalcLinearEnergiesNeon by intrinsics.
The modification only uses the unique part of the CalcLinearEnergies
 function. Pass byte to byte conformance test both on ARMv7 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6. The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |    19.48% |    19.26% |     13.68% |
| Neon inline                |    27.90% |    28.87% |     17.79% |
| Neon intrinsics (GCC 4.8)  |    18.69% |    20.18% |     14.69% |
| Neon intrinsics (LLVM 3.4) |    18.52% |    21.15% |     13.56% |

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23349004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7686 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 19:34:14 +00:00
andrew@webrtc.org
e497be3de1 replace inline assembly WebRtcAecm_StoreAdaptiveChannelNeon by intrinsics.
The modification only uses the unique part of the StoreAdaptiveChannel
 function. Pass byte to byte conformance test both on ARM32 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6.  The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |    20.97% |    37.70% |     25.41% |
| Neon inline                |    36.93% |    51.80% |     38.14% |
| Neon intrinsics (GCC 4.6)  |    27.78% |    43.71% |     26.50% |
| Neon intrinsics (GCC 4.8)  |    27.16% |    38.22% |     26.87% |
| Neon intrinsics (LLVM 3.4) |    27.82% |    39.90% |     26.69% |

Change-Id: Ia55d8a268a70164b50676c604ae40b68fc183106

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30029004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7685 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 19:32:33 +00:00
jiayl@webrtc.org
0e71070207 Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top.
BUG=crbug/403703, crbug/316603
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7684 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 18:15:55 +00:00
tnakamura@webrtc.org
a367aeab82 Bump to version 40
TBR=niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/26109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7683 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 16:23:15 +00:00
magjed@webrtc.org
f7c5d4fac7 Revert 7679 "webrtc::Scaler: Preserve aspect ratio"
> webrtc::Scaler: Preserve aspect ratio
> 
> BUG=3936
> R=glaznev@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28969004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7682 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 13:12:09 +00:00
kjellander@webrtc.org
525baea03f Add PROJECT to codereview.settings
This is needed once we move over to Chromium's
Rietveld instance at codereview.chromium.org.

BUG=3884
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7681 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 10:00:47 +00:00
kjellander@webrtc.org
944fb57140 Roll chromium_revision 375f736..91f1781
Relevant changes:
* buildtools: 51ca1f2..c27f95b
* tools/gyp: b13d8f2..487c0b6
* tools/swarming_client: 41036ec..1f8ba35
Details: 375f736..91f1781/DEPS

Clang version was not updated in this roll.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7680 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 09:57:19 +00:00
magjed@webrtc.org
809986b95f webrtc::Scaler: Preserve aspect ratio
BUG=3936
R=glaznev@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7679 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 09:51:30 +00:00
asapersson@webrtc.org
cd621a8657 Add thread annotations to overuse_frame_detector class.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 09:40:19 +00:00
henrik.lundin@webrtc.org
8038d42749 Follow-up fixes for G722
This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.

BUG=3951
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 08:38:24 +00:00
turaj@webrtc.org
1431e4dd1c Revert 7675 "Make an AudioEncoder subclass for iSAC"
Above CL did not compile on Android. Followings are links to Android builds

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320

> Make an AudioEncoder subclass for iSAC
> 
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/25019004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 01:44:13 +00:00
kwiberg@webrtc.org
05feff013e Make an AudioEncoder subclass for iSAC
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 23:53:08 +00:00
henrike@webrtc.org
33045ab2c1 Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003".
BUG=3379
R=tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/27119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7674 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 19:43:11 +00:00
henrike@webrtc.org
43e033e778 Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted."
BUG=3379
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7673 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 19:40:29 +00:00
andrew@webrtc.org
4ffc7341ca replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon by intrinsics.
The modification only uses the unique part of the ResetAdaptiveChannel
 function. Pass byte to byte conformance test both on ARM32 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6. The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |       15% |       30% |        12% |
| Neon inline                |       21% |       30% |        12% |
| Neon intrinsics (GCC 4.6)  |       19% |       32% |        12% |
| Neon intrinsics (GCC 4.8)  |       20% |       32% |        12% |
| Neon intrinsics (LLVM 3.4) |       19% |       30% |        12% |

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29019004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7672 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 17:27:53 +00:00
andrew@webrtc.org
d024f759a8 clear asm code and unused functions in audio processing module
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25119004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7671 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 17:19:57 +00:00
henrike@webrtc.org
c4922316b4 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds.
TBR=niklas.enbom@webrtc.org
BUG=3379

Review URL: https://webrtc-codereview.appspot.com/30959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7670 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 15:31:24 +00:00
pbos@webrtc.org
d819803d45 Wire up DSCP support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 14:41:43 +00:00
stefan@webrtc.org
83d4804a50 Put send-side bwe probing under finch experiment.
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7668 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 13:55:16 +00:00
pbos@webrtc.org
957e802fe0 Refactor SetDefaultEncoderConfig to work on existing codecs.
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/32009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 12:36:11 +00:00