VCM: Removing frame drop enable from Reset call
BUG = 1387

Review URL: https://webrtc-codereview.appspot.com/1097010

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3500 4adac7df-926f-26a2-2b94-8c16560cd09d
Increase maximum resolution to 4k x 3k.  
BUG=1375
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1097008

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3503 4adac7df-926f-26a2-2b94-8c16560cd09d
Sync libvpx and its gyp wrapper from Chromium.

TBR=kjellander
BUG=webrtc:1213

Review URL: https://webrtc-codereview.appspot.com/1096007

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3505 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.

TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix Windows x64 errors in video_codecs_test_framework

Fixed a few size_t converted to int warnings (interpreted as errors).
Fixed cpplint warnings.

BUG=webrtc:1323
TEST=manual compile on Windows with GYP_DEFINES=target_arch=x64 and
ninja -C out\Debug_x64 (also compiled with Release_x64)

Review URL: https://webrtc-codereview.appspot.com/1097011

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3507 4adac7df-926f-26a2-2b94-8c16560cd09d
Import SincResampler from Chromium.

Committing the originals to make further reviews cleaner.

TBR=bjornv
BUG=webrtc:1395

Review URL: https://webrtc-codereview.appspot.com/1096010

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3508 4adac7df-926f-26a2-2b94-8c16560cd09d
Import stringize_macros from Chromium.

Committing the originals to make further reviews cleaner.

TBR=bjornv
BUG=webrtc:1395

Review URL: https://webrtc-codereview.appspot.com/1106005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3509 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing lint warnings from previous commit

In this CL I have removed (almost) all lint warnings I got for this commit:
https://code.google.com/p/webrtc/source/detail?r=3454.

The only warning not fixed is a warning about usage of  non-const reference. This will be fixed in a separate CL.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1091006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3510 4adac7df-926f-26a2-2b94-8c16560cd09d
Reset ssrc when calling SetSendCodec.

BUG=1398
TEST=Tested locally.

Review URL: https://webrtc-codereview.appspot.com/1107004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3511 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove MultiStreamMode from test.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1101010

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3512 4adac7df-926f-26a2-2b94-8c16560cd09d
Added delay estimation test to audio processing unit tests.

The test verifies that we get proper delay metrics when inserting delayed versions of the same file to far-end and near-end.
Failure of the test has been verified through a missmatch between AEC delay buffer size and test buffer size.
Also added a missing file rewind to another test and removed some lint warnings.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1100004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3514 4adac7df-926f-26a2-2b94-8c16560cd09d
Moved debug file handling to aec_core from echo_cancellation.c. This removes dependency on low level member variables.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1093010

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3515 4adac7df-926f-26a2-2b94-8c16560cd09d
Roll Chromium revision 176094:182149

This gets us (for build/):
* GYP updates for Mac 64-bit builds (r178644)
* Lots of updates to Android scripts
* Support Visual Studio Express 2012.
* asan=1 now enables line numbers in symbolized ASan reports (r179326)
See
http://build.chromium.org/f/chromium/perf/dashboard/ui/changelog.html?url=trunk%2Fsrc%2Fbuild%2F&range=176094%3A182149&mode=html
for more info

In addition to this all our DEPS references to Chromium's DEPS file are
updated.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1106004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3516 4adac7df-926f-26a2-2b94-8c16560cd09d
Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1099011

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3517 4adac7df-926f-26a2-2b94-8c16560cd09d
Modify SincResampler to build in webrtc.

This is the first in a series of CLs to bring arbitrary resampling to webrtc.

* Replace Chromium-specific helpers with their respective webrtc versions.
* Add a second constructor to permit runtime selection of block_size.
* Add stringize_macros to system_wrappers.

BUG=webrtc:1395
TESTED=unit tests

Review URL: https://webrtc-codereview.appspot.com/1097012

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3518 4adac7df-926f-26a2-2b94-8c16560cd09d
Make VoiceEngineImpl inherit from VoiceEngine.
This associates the two types instead of incorrectly reinterpret casting
VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated).

Please see more details in the bug for how this is currently causing problems
with security tools.

BUG=38612
Review URL: https://webrtc-codereview.appspot.com/1099013

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
Moved the actual calculations to aec_core to avoid passing up low level members.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1103011

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3521 4adac7df-926f-26a2-2b94-8c16560cd09d
This Cl includes
* A getter for echo_state
* Style changes, such as changes to int where appropriate

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1093011

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3522 4adac7df-926f-26a2-2b94-8c16560cd09d
This refactoring CL contains an API to get low level echo metrics stats.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1107007

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3523 4adac7df-926f-26a2-2b94-8c16560cd09d
Bug fix for webrtc issue 1391. Typo in sin_length for socket address. 
Review URL: https://webrtc-codereview.appspot.com/1108004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3524 4adac7df-926f-26a2-2b94-8c16560cd09d