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Author SHA1 Message Date
fischman@webrtc.org
f61e02c81f Misc cleanups to webrtc/android code: 
- Replace some deprecated calls/enums with their more modern equivalents.
- Clean up some usage of global data and/or hide it better
- Catch specific exceptions instead of Exception, and log the exception instead
  of just its message.
- Random log message cleanups
- Added a build_with_libjingle gyp variable to mimic build_with_chromium for
  when webrtc is built as part of a libjingle project but not part of chromium.

BUG=webrtc:1169
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1105010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 23:13:46 +00:00
bjornv@webrtc.org
56a9ec30e9 Refactoring AEC: AecCore struct made private 
* Added aec_core_internal.h for private variables.
* Moved aec_t struct to aec_core_internal.h
* Name change aec_t -> AecCore
* Moved additional declarations to aec_core_internal.h
* Tested with audioproc_unittest and trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1117004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 22:38:47 +00:00
bjornv@webrtc.org
71e91f3b64 Refactor AEC: PowerLevel 
* Style changes
* Tested with audioproc_unittest and trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1116005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 19:24:50 +00:00
bjornv@webrtc.org
4d1cfae622 Added a pointer getter to the system_delay variable. 
Tested with audioproc_unittest, trybots

TEST=None
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1101015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 17:31:38 +00:00
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
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
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
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
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
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
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
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
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
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
henrikg@webrtc.org
b64732abfc Fix Win64 build breakage 
This is for landing https://webrtc-codereview.appspot.com/1096006/ by Justin Schuh.

Stable will be updated after this has landed.

Review URL: https://webrtc-codereview.appspot.com/1091008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-07 10:14:05 +00:00
kma@webrtc.org
d83b9fdf45 Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged. 
Bugs=none
Test=trybots, and file bit-exact tests; passed.

Description of the bug: Neon registers q4-q7 not saved before calling the outside FFT routines in the assembly functions.
Review URL: https://webrtc-codereview.appspot.com/1097006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-06 23:53:13 +00:00
kma@webrtc.org
959da8d286 Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.). 
Bugs: none
Review URL: https://webrtc-codereview.appspot.com/1072007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-05 20:46:55 +00:00
phoglund@webrtc.org
a7303bdfb5 Lint-cleaned video and audio receivers. 
BUG=
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/1093004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-05 15:12:39 +00:00
phoglund@webrtc.org
244251a9cd Moved almost all payload-related stuff to the payload registry. 
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.

BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1078004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 13:23:07 +00:00
kjellander@webrtc.org
fa53d8717c Fixing/disabling Windows x64 warnings 
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.

With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.

BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64

Review URL: https://webrtc-codereview.appspot.com/1060008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 10:07:17 +00:00
braveyao@webrtc.org
254d85af54 Exchange TRY by enumerating image formats in Linux video capture 
ISSUE = issue 529
TEST  = unittest on Linux
Review URL: https://webrtc-codereview.appspot.com/1066011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 07:53:53 +00:00
stefan@webrtc.org
becf9c897c Fix mismatch between different NACK list lengths and packet buffers. 
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.

BUG=1289

Review URL: https://webrtc-codereview.appspot.com/1065007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 15:09:57 +00:00
stefan@webrtc.org
b586507986 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. 
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.

BUG=1298

Review URL: https://webrtc-codereview.appspot.com/1060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 14:33:42 +00:00
tina.legrand@webrtc.org
46d90dcd74 Adding three frame sizes to Opus 
Adding support for 10, 40 and 60 ms packet sizes for Opus.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1086004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 14:20:06 +00:00
henrik.lundin@webrtc.org
aaad6134b9 Implementing stereo support for G.722 
This CL implements stereo support for G.722 through a new class
AudioDecoderG722Stereo derived from AudioDecoderG722.

Also implementing tests for G.722 stereo.

Review URL: https://webrtc-codereview.appspot.com/1073006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 11:49:28 +00:00
bjornv@webrtc.org
ac46c6dac3 Replaced relative path to reference from <(webrtc_root). 
Changed to proper include paths in AECM and NSX.
Tested on trybots.

BUG=None

Review URL: https://webrtc-codereview.appspot.com/1063014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 21:06:16 +00:00
turaj@webrtc.org
763faeab4e Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. 
This is discovered during a test for controlling delay. It is not simple to reproduce it. 

Bug=
test=manual test verified that |functionDurationEst| is correctly updated.
Review URL: https://webrtc-codereview.appspot.com/1074013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 18:21:58 +00:00
turaj@webrtc.org
c0ada864b2 fix for issue 281. 
A reverse copy is removed. The index to src buffer could be -1, this happens very often. The reverse copy is not needed as the content of the destination is overwritten further down in "WebRtcIlbcfix_CbConstruct()" 


Bug=issue281
TEST=manual test over 1600 files TIMIT database, all outputs are bit-exact with the ones generated from head revision. Local run of asan does not generate any warning.
 
Review URL: https://webrtc-codereview.appspot.com/1063013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 18:21:06 +00:00
mikhal@webrtc.org
119c67df36 Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value. 
This cl also includes tests and some clean up.

Review URL: https://webrtc-codereview.appspot.com/1019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 17:18:02 +00:00