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1309 Commits

Author SHA1 Message Date
bemasc@google.com
603ae3ece2 Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. 
This CL changes the return code of these methods to indicate
success instead of failure when there is nothing to change.

This change appears to resolve an issue where enabling the
timestamp offset extension via SDP would result in a failure if
that extension had already been enabled.
Review URL: https://webrtc-codereview.appspot.com/1118008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 17:03:02 +00:00
andrew@webrtc.org
78693fe37c Return an error when greater than 16 kHz is used with AECM. 
BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1146005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 16:36:19 +00:00
marpan@webrtc.org
7d052c3cb2 Turn off error concealment in videoprocessor_integration tests. 
Review URL: https://webrtc-codereview.appspot.com/1123006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 16:44:28 +00:00
braveyao@webrtc.org
6b6eb44cca Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. 
ISSUE=529
TEST=unittest
Review URL: https://webrtc-codereview.appspot.com/1120006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 10:08:02 +00:00
stefan@webrtc.org
9e254133ad Rewrite the jitter buffer statistics test and put make it robust under valgrind. 
BUG=1158

Review URL: https://webrtc-codereview.appspot.com/1116008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 08:45:23 +00:00
bjornv@webrtc.org
132c15de30 AEC Refactoring: 
* Adds pointer to low level AecCore struct.
* Adds a simple unit test of this new call.

Tested with audioproc_unittest, trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1121006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 21:03:41 +00:00
stefan@webrtc.org
e1c4ed958d Fix to send a full NACK list at least roughly once every 1.5 x RTT. 
BUG=1434

Review URL: https://webrtc-codereview.appspot.com/1111007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 16:23:06 +00:00
kma@webrtc.org
83561fb173 Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. 
No written bug report.

Tested with Nexus-S. Issue disappeared with the change.
Review URL: https://webrtc-codereview.appspot.com/1126006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 01:16:44 +00:00
andrew@webrtc.org
91f325586d Refactor WebRtc_CreateBuffer to return the instance. 
Review URL: https://webrtc-codereview.appspot.com/1140005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 00:35:06 +00:00
andrew@webrtc.org
9fbd9ca849 Force a memcpy directly from the AEC ring buffer. 
Review URL: https://webrtc-codereview.appspot.com/1140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 23:31:56 +00:00
andrew@webrtc.org
ac1f877a5e Remove unneeded libvpx path from vp8 include_dirs. 
This is already provided by libvpx.gyp.

BUG=webrtc:1428

Review URL: https://webrtc-codereview.appspot.com/1139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 17:10:38 +00:00
andrew@webrtc.org
9ae1354e25 Refactor ring_buffer interface, add a feature and a test. 
* Add a RingBuffer typedef.
* Add the ability to force a memcpy by passing a null ptr. In some cases,
  we know we want a memcpy. This allows us to skip a potential
  intermediate memcpy.
* Add a stress test.

Review URL: https://webrtc-codereview.appspot.com/1111004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 17:07:35 +00:00
phoglund@webrtc.org
8a0662306d New attempt at fixing hard-coded libvpx source. 
TBR=ajm@webrtc.org
BUG=1428

Review URL: https://webrtc-codereview.appspot.com/1138004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 15:05:01 +00:00
phoglund@webrtc.org
9a6623b629 Revert "Fixing hard-coded libvpx source path." 
This reverts commit 1c603646da11971f13d66a75da31ebbb6eff37d9.

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1137004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 14:27:14 +00:00
phoglund@webrtc.org
8571c90c7f Fixing hard-coded libvpx source path. 
BUG=1428
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1127006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 14:09:12 +00:00
kma@webrtc.org
2f9bd247ad Ported assembly coding in APM from Android to iOS. 
Bugs=none
Test=trybots, and offline file bit-exact tests.
Review URL: https://webrtc-codereview.appspot.com/1066009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-23 04:16:59 +00:00
vikasmarwaha@webrtc.org
10987a851d Minor bug fix in maxFPS parameter declaration. 
TBR = perkj@webrtc.org,mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1123005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3559 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 22:03:39 +00:00
vikasmarwaha@webrtc.org
bf3a9b3cce Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero. 
Review URL: https://webrtc-codereview.appspot.com/1101013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 20:25:54 +00:00
bjornv@webrtc.org
60f83131e4 AEC refactoring: Moved typedefs to _internal.h 
* This was actually part of r3553
* Tested with audioproc_unittest, trybots

TBR=andrew@webrtc.org
TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1118005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 16:12:24 +00:00
tina.legrand@webrtc.org
7a7a008031 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

Committed: https://code.google.com/p/webrtc/source/detail?r=3543

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 10:27:48 +00:00
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