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Author SHA1 Message Date
bjornv@webrtc.org
47b274de44 Refactoring AEC: Added a SetConfigCore function 
* Configuraion parameters now passed down the AEC Core struct.
* Tested with audioproc_unittest and on trybots.

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1098014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3548 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 17:09:47 +00:00
bjornv@webrtc.org
716fd90ff2 Moved out buffer handling to ProcessFrame() 
Tested with audioproc_unittest, trybots and verified bit exactness on recording data base.

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1110006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3547 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 16:59:41 +00:00
bjornv@webrtc.org
ee7202f7a4 Removed unused get_config function. The configuration is already stored and handled in the audio processing module, so there is no need for this functionality. 
Tested with audioproc_unittest and on trybots.

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1103016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3546 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 16:36:59 +00:00
mflodman@webrtc.org
59b2d5fbce Stop and restart fix. 
BUG=1398
TEST=Local stop and start test.

Review URL: https://webrtc-codereview.appspot.com/1115004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3545 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 16:00:27 +00:00
tina.legrand@webrtc.org
eb7ebf20ed Revert 3543 
> Changing non-const reference arguments to pointers, ACM
> 
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
> 
> BUG=issue1372
> 
> Review URL: https://webrtc-codereview.appspot.com/1103012

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:57:31 +00:00
tina.legrand@webrtc.org
374aa49e1a Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:22:23 +00:00
pbos@webrtc.org
0b6293aaaa Fixed typo in vie_autotest_loopback.cc. 
Review URL: https://webrtc-codereview.appspot.com/1114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 12:13:10 +00:00
andrew@webrtc.org
83663efba4 Replace gtest_prod.h include with our own FRIEND_TEST macro. 
This small bit of duplication avoids depending on any part of GTest in
production code.

TBR=phoglund
BUG=1395

Review URL: https://webrtc-codereview.appspot.com/1098013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 23:43:57 +00:00
fischman@webrtc.org
aea96d36e3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver 
to avoid ODR violations with peerconnectioninterface.h in libjingle.

Conflict introduced in
https://webrtc-codereview.appspot.com/1060005/diff/14010/webrtc/modules/interface/module_common_types.h#newcode326

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1105011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 22:09:36 +00:00
bjornv@webrtc.org
0a480cbe4d Added getter for far_time_buf in AEC. Only used in AEC debug dump. 
TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1110005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 21:41:27 +00:00
bjornv@webrtc.org
5fc829200c This refactoring CL moves the nlp_mode member value from aecpc_t to aec_t, since it it never used at that level. Further, I removed two suppression variables by depending on nlp_mode directly. 
TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1095007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 21:06:52 +00:00
bjornv@webrtc.org
cea70f4055 * Name change 
* Removed WebRtcAec_ function name prepending on private function.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1096012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 21:03:10 +00:00
marpan@webrtc.org
95b48c3551 Update to codec unit test: 
enable frame dropper for rate control test.
Review URL: https://webrtc-codereview.appspot.com/1099014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 20:02:32 +00:00
mikhal@webrtc.org
77fced32e2 fixing nack list size calculation 
Review URL: https://webrtc-codereview.appspot.com/1093012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 19:42:35 +00:00
elham@webrtc.org
10741b32b8 Updated version number to 3.24 
Review URL: https://webrtc-codereview.appspot.com/1110004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 19:05:14 +00:00
mikhal@webrtc.org
1682f71210 Updating watchlist 
Review URL: https://webrtc-codereview.appspot.com/1101012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 18:31:20 +00:00
phoglund@webrtc.org
ba23d11564 Will now run pylint on all python files if there's at least one modified python file in the checkin. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1101011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3531 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 12:27:57 +00:00
tommi@webrtc.org
0460c7294a Remove the dependency on dxguid.lib. 
It turns out we don't really need it and therefore can also get rid of the added lib directory.
Review URL: https://webrtc-codereview.appspot.com/1094015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 12:13:03 +00:00
tommi@webrtc.org
d2c3bed1da Move directx_sdk_path definition variable into the video_render_module gyp file. 
The variable is now:
* Only set and used for Windows (not globally for all platforms)
* Only used in the standalone build (include_internal_video_render == 1)

This means that we can remove the variable from Chrome and that the standalone
win builders should start picking up the local directx folder and turn green
(*crossesfingers*).
Review URL: https://webrtc-codereview.appspot.com/1103014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-18 14:53:04 +00:00
stefan@webrtc.org
eb91792cfd Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. 
Review URL: https://webrtc-codereview.appspot.com/1105007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-18 14:40:18 +00:00
mikhal@webrtc.org
3897255b63 Add VoE interface to VieRTP test 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1097015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-16 01:35:59 +00:00
marpan@webrtc.org
e3d6ffede4 Increase threshold in codec unit test. 
Review URL: https://webrtc-codereview.appspot.com/1096011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3526 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 23:31:17 +00:00
mikhal@webrtc.org
ef9f76a59d Adding a receive side API for buffering mode. 
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 23:22:18 +00:00
vikasmarwaha@webrtc.org
47fe5736c1 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@3524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 18:42:12 +00:00
bjornv@webrtc.org
b4cd342eb9 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@3523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 18:40:34 +00:00
bjornv@webrtc.org
21a2fc902d 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@3522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 17:01:03 +00:00
bjornv@webrtc.org
325f625137 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@3521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 15:21:02 +00:00
tommi@webrtc.org
0989fb7bfa 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@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 15:07:32 +00:00
phoglund@webrtc.org
17238576ba Removed astyle from webrtc_reformat since clang-format-chrome.py handles that now. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1101009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 09:43:20 +00:00
andrew@webrtc.org
076fc12539 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@3518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 03:54:22 +00:00
bjornv@webrtc.org
6f6acd9f80 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@3517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 21:17:12 +00:00
kjellander@webrtc.org
4013ac478e 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@3516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 19:13:30 +00:00
bjornv@webrtc.org
7267ffde56 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@3515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 17:56:23 +00:00
bjornv@webrtc.org
3e10249f20 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@3514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 15:29:09 +00:00
kjellander@webrtc.org
e580be993c Add regression monitoring for audioproc and iSAC fixed-point tests. 
BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1094011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 12:27:17 +00:00
stefan@webrtc.org
07b667db5e 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@3512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 11:35:20 +00:00
mflodman@webrtc.org
294e5b0b82 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@3511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 11:25:26 +00:00
tina.legrand@webrtc.org
a092cbf9b7 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@3510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 09:28:10 +00:00
andrew@webrtc.org
45eab19e7d 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@3509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-13 23:37:14 +00:00
andrew@webrtc.org
a8ef811fe5 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@3508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-13 23:00:49 +00:00
kjellander@webrtc.org
9c4e662ea8 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@3507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-13 09:35:12 +00:00
turaj@webrtc.org
6388c3e2fd 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@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 21:42:18 +00:00
andrew@webrtc.org
e6e344a7dc 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@3505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 19:35:18 +00:00
fbarchard@google.com
0ee57c2436 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@3503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 04:57:56 +00:00
mikhal@webrtc.org
57a0049e25 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@3500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 21:23:23 +00:00
kjellander@webrtc.org
18a21a03c6 Android NDK build tools 
This CL enables building with Android NDK in the way that Chromium buildbots do it.

== Overview ==
* Add Android dependencies to DEPS (SDK, NDK, Android test runner). This also makes it possible to use Android's build/android/run_tests.py script to execute tests on Android devices.
* Add a Python script to build the WebRTC Video demo for Android using ndk-build and Ant. This is designed as an annotation script for Buildbots but is also fine to run locally.
* Update Android.mk so it works with the compiler output from a build performed by build/android/buildbot/bb_run_bot.py (which is how Chrome buildbots build).

== Syncing Android dependencies ==
To get the dependencies added in DEPS synced out, you must change the last line
of your .gclient file to look like this:
];target_os = ["android"]

That will append another variable to the .gclient file that causes these
dependencies to be synced during gclient sync.
If you want to get additional platform-specific dependencies in the same
checkout, add them to the list too, e.g. target_os = ["android", "unix"].

== Android.mk ==
The fix in Android.mk is needed since Chrome is building using build/android/buildbot/bb_run_bot.py, which only output the libraries into out/Debug. With the change it works for both that and a normal build (which copies the library files from out/Debug/obj.target/subpath to out/Debug anyway as a part of the build).

== svn:ignore ==
NOTICE: Before submitting, the following directories should be added to svn:ignore in third_party to avoid them from being removed and re-synced for every build:
* android_testrunner
* android_tools
* WebKit
This has to be done in a manual SVN commit since it's not possible to include in a git-svn CL (and I don't want to migrate this to a SVN CL).

BUG=none
TEST=local builds

Review URL: https://webrtc-codereview.appspot.com/1024009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 17:43:19 +00:00
kjellander@webrtc.org
00ab7cf4fd Fix perf output for audioproc and iSAC fixed-point tests 
The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.

BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1093007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 12:33:03 +00:00
stefan@webrtc.org
0cb48a0a18 Set SingleStream BWE in unittests. 
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1094004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 08:30:23 +00:00
stefan@webrtc.org
63066f7200 Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage. 
TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1098010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 08:27:33 +00:00
mikhal@webrtc.org
3d305c64b4 Updates to send side streaming mode: 
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.

Review URL: https://webrtc-codereview.appspot.com/1101007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-10 18:42:55 +00:00