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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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