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464 Commits

Author SHA1 Message Date
mikhal@webrtc.org
80dd19be0a vplib tests: Removing old and unused file and directories. 
Note that the convert_test and scale_test directories are also removed. 
Review URL: http://webrtc-codereview.appspot.com/208001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 22:57:06 +00:00
henrike@webrtc.org
bf54ef9bb7 Removed code under a non-existing define. 
Review URL: http://webrtc-codereview.appspot.com/193006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 18:14:25 +00:00
henrike@webrtc.org
1a2933c71a Fixes a Valgrind warning triggering when the number of pending messages hit the limit. 
Review URL: http://webrtc-codereview.appspot.com/200002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 17:55:56 +00:00
andrew@webrtc.org
b2d4921f3b Remove trailing whitespace in AudioDevice. 
(That I introduced...)
Review URL: http://webrtc-codereview.appspot.com/198002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 16:34:36 +00:00
mikhal@webrtc.org
d6132f54d2 Review URL: http://webrtc-codereview.appspot.com/193007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 16:23:38 +00:00
kjellander@webrtc.org
35a1756502 First version of video quality measurement program and test framework. 
See https://docs.google.com/a/google.com/document/d/1w6Nrxw6yTg_sDu18Ux8oZPEMo5F_R-zt62udrmmTeOc/edit?hl=en_US
for background, details and additional instructions on usage.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/175001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 06:44:54 +00:00
andrew@webrtc.org
3ce62fcfe4 Move merge_libs targets to their own gyp. 
The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries.
Review URL: http://webrtc-codereview.appspot.com/191003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 01:03:18 +00:00
kma@webrtc.org
af57de006a Some code style changes in audio_processing/ns/main/source/ by Astyle, 
with a little manual modification.
Review URL: http://webrtc-codereview.appspot.com/201002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 23:36:01 +00:00
henrik.lundin@webrtc.org
01ca01f6e6 Adding neteq_tests to modules tests 
Also moving neteq_tests.gyp and renaming to gypi. Cleaning up a
little in neteq_tests.gypi.

Review URL: http://webrtc-codereview.appspot.com/191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 20:38:19 +00:00
kma@webrtc.org
bbc1f10187 Changed modules/audio_processing/utility/Android.mk, to correct a build error in 
Android with the change from version r674.
Review URL: http://webrtc-codereview.appspot.com/197003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 18:09:02 +00:00
kma@webrtc.org
bf39ff4271 Some general optimization in NS. 
No big effort in introducing new style.
Speed improved ~2%.
Bit exact.
Will introduce mulpty-and-accumulate and sqrt_floor next, which increase speed another 2% or so.

Note: In function WebRtcNsx_DataAnalysis, did the block separation because I found one "if" case is more frequent than "else" within a for loop; rest is kind of code re-aligning.
Review URL: http://webrtc-codereview.appspot.com/181002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 17:10:06 +00:00
kma@webrtc.org
a58224f9f0 Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7. 
It's used in quite some occations over many modules.
Review URL: http://webrtc-codereview.appspot.com/178004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 16:44:11 +00:00
stefan@webrtc.org
4b6f747373 Fixes a newly introduced bug in the jitter buffer where buffer reallocation 
causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f Fixed bug in jitter buffer which caused the missingFrames bit to never be set. 
Also updated the VP8 wrapper to return fully concealed frames (for rendering).

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/190003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:48:11 +00:00
stefan@webrtc.org
61b4abf1f8 Proper use of frame rate argument in generic_codec_test. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:40:21 +00:00
mikhal@webrtc.org
e06be4f678 video coding tests: Adding ssimFrame to interface 
Review URL: http://webrtc-codereview.appspot.com/188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:43 +00:00
mikhal@webrtc.org
ae7a0522c5 video_coding robustness: Updating hybrid mode's settings 
1. Disabling adjustment factor - temporary update. 
2. Enabling a windowed filtered loss for the hybrid mode.  
Review URL: http://webrtc-codereview.appspot.com/192003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:34 +00:00
marpan@google.com
f1f3fb33b5 Update to rate-mismatch factor in media_opt_util. 
Review URL: http://webrtc-codereview.appspot.com/193003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 19:09:45 +00:00
andrew@webrtc.org
f458916145 Returning errors if any of the Init() settings in VoE fail. 
There's no reason to try to continue if these simple settings fail; better to know about it immediately.

Also, readjusting the indentation to avoid breaking strings over several lines. This bends GStyle a bit, but it's well worth it to avoid the common "forgot to add a space" error.
Review URL: http://webrtc-codereview.appspot.com/173003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 15:22:28 +00:00
stefan@webrtc.org
5b91464edf Allow an aggregated partition to spill over to a new packet. 
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 10:26:12 +00:00
bjornv@google.com
1ba3dbecbb Adds possibility to log delay estimates in AEC. 
Review URL: http://webrtc-codereview.appspot.com/178001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 08:18:10 +00:00
stefan@webrtc.org
f72c36763f Reverting changelist 666 since it broke the build on Mac. 
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/187003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 07:37:41 +00:00
andrew@webrtc.org
6d169f2474 Fix Mac build error in vie_auto_test introduced in r666. 
COCOA_RENDERING was undefined. Committing without review.
Review URL: http://webrtc-codereview.appspot.com/191002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 06:00:42 +00:00
mflodman@webrtc.org
5eec6cf29a Started rewriting video_engine tests to use GUnit. 
- Added comments to the new test.
- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/168002
Patch from Patrik Hoglund <phoglund@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 12:24:13 +00:00
punyabrata@webrtc.org
6b6d08164f Remove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected. 
Review URL: http://webrtc-codereview.appspot.com/180001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 17:45:03 +00:00
kma@google.com
c611b1a950 Bit-exact with non-Neon version. 
Review URL: http://webrtc-codereview.appspot.com/180002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 16:03:38 +00:00
bjornv@google.com
0beae6798d Removed level estimator calls, since it is not supported. There are still one place left; used within SetRTPAudioLevelIndicationStatus(). The error return value of level_estimator() has no effect there. 
The VoE auto tests have been updated as well.
Review URL: http://webrtc-codereview.appspot.com/178003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 14:08:19 +00:00
andrew@webrtc.org
18421f2063 Remove unnecessary include from NS interface. 
http://code.google.com/p/webrtc/issues/detail?id=46
Review URL: http://webrtc-codereview.appspot.com/183001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:50:52 +00:00
amyfong@webrtc.org
6a23ad5702 Fixed the CameraCap button to say Version, also change the function name inside ChannelDlg.cpp 
Review URL: http://webrtc-codereview.appspot.com/182001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:19:10 +00:00
amyfong@webrtc.org
2d08d43206 * Added modification of Start Bit Rate to vie_auto_test_custom_call 
* Added minor spacing and ":" for user input during vie_auto_test_custom_call
* Changed the default Video Port to 11111 and Audio Port to be 11113 to bring it inline with the WindowsTest application for ViE
Review URL: http://webrtc-codereview.appspot.com/181001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 17:46:45 +00:00
mikhal@webrtc.org
848fad23c6 video_coding: Updating media opt test - fixing call to protection callback. 
Review URL: http://webrtc-codereview.appspot.com/179003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 16:30:59 +00:00
xians@google.com
49d025f262 Get the right guid str for GetRecordingDeviceName 
Bug=http://code.google.com/p/webrtc/issues/detail?id=99
Test=none
Review URL: http://webrtc-codereview.appspot.com/183002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 14:43:06 +00:00
bjornv@google.com
a2c6ea09b0 Removed a segmentation fault error when processing near_file only. 
Review URL: http://webrtc-codereview.appspot.com/174001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 08:04:45 +00:00
kma@google.com
961885a8bb In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7. 
Review URL: http://webrtc-codereview.appspot.com/140010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-26 16:35:25 +00:00
mikhal@webrtc.org
e185e9f68a video_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list. 
Review URL: http://webrtc-codereview.appspot.com/165001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 22:02:40 +00:00
turajs@google.com
cf136186f5 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:49:25 +00:00
turajs@google.com
13335ccd7a Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:47:25 +00:00
turajs@google.com
610f478705 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:45:34 +00:00
turajs@google.com
53439d9982 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:44:00 +00:00
amyfong@webrtc.org
713f91e12b Fixed vie_autotest_custom_call.cc minor issues. 
1. mirror of local render removed
2. the video device the user selected wasn't what was actually being used when the call is being made
3. fixed mentions of loopback calls
Review URL: http://webrtc-codereview.appspot.com/171001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:26 +00:00
mikhal@webrtc.org
105ff39dec video coding: updating offline tests. 
Additional clean-up to the offline test: Placing test callbacks in a designated file. 
Review URL: http://webrtc-codereview.appspot.com/167002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:11 +00:00
turajs@google.com
496ef8aca8 To fix warnings in test files. 
Review URL: http://webrtc-codereview.appspot.com/169001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 15:45:48 +00:00
bjornv@google.com
8e9e83b530 This CL adds guards against division by zero, that should fix http://b/issue?id=5278531 
In addition a read outside memory event has been detected and removed.
Also an improper noise weighting has been corrected.
Review URL: http://webrtc-codereview.appspot.com/152001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 12:39:47 +00:00
kjellander@webrtc.org
9e7774f163 Added compare methods for TickInterval class. 
This is useful to be able to sort them using the STL algorithm library.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/173002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 11:33:31 +00:00
bjornv@google.com
dc743a8bba Replaces a use of log2. 
I've replaced a log2 operation so it works on Windows.
Review URL: http://webrtc-codereview.appspot.com/171002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 08:13:53 +00:00
leozwang@google.com
90eff6c7c6 Fix compilation error in build-in AEC test 
Review URL: http://webrtc-codereview.appspot.com/164001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 18:02:03 +00:00
wu@webrtc.org
221b522118 Return the number of /dev/video* without trying to open it. 
Consider the case when there're /dev/video0 and /dev/video1. But for somereason the video0 is not in a correct state and can't be open. As a result, current NumberOfDevices will return 1, which is fine. However, we will then never be able to get the device we really want - /dev/video1. Consider the code below, the GetCaptureDevice will fail because it calls into DeviceInfoLinux::GetDeviceName(0, ...) which will again try to open the /dev/video0. So the root cause is the mismatching of the NumberOfDevices and GetDeviceName.

Since we will open the device in DeviceInfoLinux::GetDeviceName anyway, I think we should return the number of /dev/video* in DeviceInfoLinux::NumberOfDevices without trying to open it. Otherwise the DeviceInfoLinux::NumberOfDevices should return more information like which /dev/video* is valid which is not.

bool found = false;
for (int i = 0; i < vie_capture->NumberOfCaptureDevices(); ++i) {
  if (vie_capture->GetCaptureDevice(i, ...) == 0) {
    found = true;
    break;
  }
}
Review URL: http://webrtc-codereview.appspot.com/148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 16:57:15 +00:00
bjornv@google.com
65e6ab31eb Temporary log2 remove to build in chrome 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 11:56:46 +00:00
amyfong@webrtc.org
3be70ca17e Added mute, hold and typing detect to voe_cmd_test to increase functionality in the voe_cmd_test application. 
Typing Detect is applicable only for Mac.  
Review URL: http://webrtc-codereview.appspot.com/156002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 23:41:06 +00:00
wu@webrtc.org
a1930427af When WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. 
Review URL: http://webrtc-codereview.appspot.com/167001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 17:38:57 +00:00
leozwang@google.com
657f483c26 Fix compilation error 
Review URL: http://webrtc-codereview.appspot.com/162003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@630 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 16:41:20 +00:00
leozwang@google.com
ec5e87614e Enable OPENELSE defination when compile voice engine 
Review URL: http://webrtc-codereview.appspot.com/150005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 16:41:09 +00:00
pwestin@webrtc.org
741da942ec Added support for new RTCP message REMB (remote estimated max bitrate) 
Review URL: http://webrtc-codereview.appspot.com/149001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 13:52:04 +00:00
andrew@webrtc.org
86b85db67e Add missing intrinsic casts for VS 2005. 
Allows re-enabling SSE optimization on Windows.
Review URL: http://webrtc-codereview.appspot.com/161003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 18:48:25 +00:00
leozwang@google.com
522f42bb80 Add kPlatformAndroid to platform check function 
Review URL: http://webrtc-codereview.appspot.com/161002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 17:39:05 +00:00
andrew@webrtc.org
ed083d4079 Modify the _vadActivity member of the AudioFrame passed to AudioProcessing. 
This saves the user from having to explicitly check stream_has_voice(). It will allow typing detection to function, which relies on this behaviour.
Review URL: http://webrtc-codereview.appspot.com/144004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 15:28:51 +00:00
andrew@webrtc.org
94c7413b0d Allow echo metrics to be enabled in process_test. 
Review URL: http://webrtc-codereview.appspot.com/155002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 15:17:57 +00:00
henrik.lundin@webrtc.org
4c36d3b424 Fixing windows warnings in rtp_utility 
Adding explicit casting to bool to avoid warnings when compiling
in windows.

Review URL: http://webrtc-codereview.appspot.com/140002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 08:16:20 +00:00
andrew@webrtc.org
67812a4621 Temporarily disabling SSE2 on Windows again until we can build on VS 2005. 
Skipping review because the build is broken on Windows.
Review URL: http://webrtc-codereview.appspot.com/156003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 02:28:49 +00:00
andrew@webrtc.org
d02dc6e682 Removing bwe_standalone from modules.gyp 
Review URL: http://webrtc-codereview.appspot.com/144003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-17 00:44:23 +00:00
wjia@google.com
fdaee9c014 include build/common.gypi directly 
Review URL: http://webrtc-codereview.appspot.com/153006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-17 00:06:08 +00:00
andrew@webrtc.org
7b7c045b75 Fix MSVC issues in AEC to enable SSE2 optimization on Windows. 
Variables now declared at top of scope and replacing C casts with intrinsic cast functions.
Review URL: http://webrtc-codereview.appspot.com/160001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 22:51:57 +00:00
leozwang@google.com
b37ec71dbd Remove delay_estimator_float.c from android build 
Review URL: http://webrtc-codereview.appspot.com/161001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 21:50:36 +00:00
leozwang@google.com
a5700876c0 Add include path to auto test 
Review URL: http://webrtc-codereview.appspot.com/155001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 20:38:31 +00:00
andrew@webrtc.org
416d702ace Fix autotest error on non-Win platforms. 
Review URL: http://webrtc-codereview.appspot.com/149007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 14:22:39 +00:00
leozwang@google.com
ce95069ade Fix buidling error 
Review URL: http://webrtc-codereview.appspot.com/151002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 22:28:08 +00:00
wu@webrtc.org
fcd12b3b7d Add necessary spaces to log. 
Review URL: http://webrtc-codereview.appspot.com/148002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 20:49:50 +00:00
andrew@webrtc.org
4537c2a464 Remove the UNCONSTR code path from AEC. 
Leave the unconstrained filter adaptation in a commented out function. Consider using this for a low-complexity mode.
Review URL: http://webrtc-codereview.appspot.com/146001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 18:25:14 +00:00
tommi@webrtc.org
8dc3985a10 Fix windows build. 
Review URL: http://webrtc-codereview.appspot.com/150001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 15:01:04 +00:00
bjornv@google.com
b47d4b287d This CL includes a move of the fixed point delay estimator from aecm to apm/utility. There has also been a code change that makes it possible to enable/disable the far end alignment, so that we save complexity when used as a quality metrics. 
Review URL: http://webrtc-codereview.appspot.com/135014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 12:27:36 +00:00
henrik.lundin@webrtc.org
29fd9a5f30 Removing warnings in all NetEQ test targets 
Now all targets in neteq.gypi builds again. Also added payload type to
the log produced by RTPanalyze.

Review URL: http://webrtc-codereview.appspot.com/148001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 08:25:45 +00:00
andrew@webrtc.org
19eefdc9f0 Add a unit testing framework. 
Populate it with the beginnings of a resampler unit test to have it do someting.

Also fix a bug in resampler caught with the test ;)
Review URL: http://webrtc-codereview.appspot.com/135019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-14 17:02:44 +00:00
andrew@webrtc.org
b524f441d0 Correct some comment spelling errors. Skipping review. 
Review URL: http://webrtc-codereview.appspot.com/144002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 18:04:30 +00:00
andrew@webrtc.org
7a585a7903 Correct voe_auto_test file path on Windows. 
Needs to be changed due to the recent move of voice_engine.gyp.
Review URL: http://webrtc-codereview.appspot.com/144001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:46:36 +00:00
andrew@webrtc.org
a3c6d61c44 Integrate the built-in WASAPI AEC DMO to VoE. 
Review URL: http://webrtc-codereview.appspot.com/108006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:17:49 +00:00
leozwang@google.com
b1b3e67c97 Fix compilation errors 
Review URL: http://webrtc-codereview.appspot.com/142002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:16:24 +00:00
leozwang@google.com
803a5f2795 Add include path 
Review URL: http://webrtc-codereview.appspot.com/141003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:15:22 +00:00
andrew@webrtc.org
2cef36fa98 Fix Windows gyp run. 
On Windows, gyp seems to require valid source files. The matlab_plotting_test target was missing its one source file, so I removed the target.

Also moving bwe_standalone.gypi to the test include list.
Review URL: http://webrtc-codereview.appspot.com/143001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:03:15 +00:00
andrew@webrtc.org
f5fb095bf9 Fix audio processing tests gypi after recent changes. 
Review URL: http://webrtc-codereview.appspot.com/137025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 01:04:59 +00:00
andrew@webrtc.org
938a049dbe Clean up vie_auto_test.gypi. 
The most significant change is to use gyp's "copies" feature for copying the media files. This will now only trigger when needed, rather than with every build as before.
Review URL: http://webrtc-codereview.appspot.com/133020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 22:06:38 +00:00
marpan@google.com
45fa141f0a qm_select: changed default settings for uep. 
Review URL: http://webrtc-codereview.appspot.com/132015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 16:53:19 +00:00
henrik.lundin@webrtc.org
9f710d08e1 Switch to new sqrt in NetEQ 
Switched to WebRtcSpl_SqrtFloor instead of WebRtcSpl_Sqrt in
NetEQ. The output is not bit-exact, but subjective listening
tests show no audible difference. Analysis shows that almost
all of the difference is in changed delay.

The reference file for NetEQ's unit test was updated.

Review URL: http://webrtc-codereview.appspot.com/139019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 16:44:37 +00:00
kjellander@webrtc.org
f0a8464b74 Added more statistics during SSIM/PSNR calculation, including calculation of min/max value. 
Moved video_metrics.h into a GYP library so it can be used from other projects.

Review URL: http://webrtc-codereview.appspot.com/132013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 13:45:39 +00:00
xians@google.com
d3185fe219 refactor the gyp file to gypi file. 
Basically, the gypi file is a copy of gyp file, but has some difference on the
path of the dependencies.
Review URL: http://webrtc-codereview.appspot.com/137020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 12:24:39 +00:00
perkj@webrtc.org
0cc68dc38a Change Video capture module to be reference counting. Also prevent the module from beeing deleted using the interface. 
Furthermore remove all static module creation and deletion functions.
Review URL: http://webrtc-codereview.appspot.com/133012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 08:53:36 +00:00
tina.legrand@webrtc.org
31c6b60456 Adding calls to Version functions for external codecs. 
Also clarified in comments where to put interface files for external codecs.
Review URL: http://webrtc-codereview.appspot.com/135017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 07:18:37 +00:00
zakkhoyt@google.com
b448ae229c Permanently adding additional logs 
Review URL: http://webrtc-codereview.appspot.com/137024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:41:49 +00:00
zakkhoyt@google.com
c6e8b72c83 Removing qualifiers on include path 
Review URL: http://webrtc-codereview.appspot.com/132014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:41:13 +00:00
marpan@google.com
30ecda146a media_opt_util: Added comment and lowered window size parameter. 
Review URL: http://webrtc-codereview.appspot.com/135018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:15:12 +00:00
marpan@google.com
3f28061f3a media_opt_util: Modification to correction factor in FEC overhead. 
Review URL: http://webrtc-codereview.appspot.com/133019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:39:40 +00:00
punyabrata@webrtc.org
955d0eed2f Removing echo warning because it seems to be flooding the logs 
anytime there is any echo. Secondly, this should be treated as
a warning in the sense that echo in the signal does not mean
something is wrong with the engine.f
Review URL: http://webrtc-codereview.appspot.com/139018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:28:51 +00:00
kma@google.com
0ada410d37 Corrected conditional compilations for spl inline functions in ARM. 
Review URL: http://webrtc-codereview.appspot.com/140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:23:50 +00:00
mikhal@webrtc.org
6f54c20703 video coding test: Adding MT functionality 
Review URL: http://webrtc-codereview.appspot.com/135008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 14:38:59 +00:00
henrik.lundin@webrtc.org
35dcc23110 Adding regression test to NetEQ 
The test inputs RTP packets from an RTPdump file into NetEQ
and compares the output to the corresponding reference file.
Test files are included.

The change also includes a new method in NETEQTEST_RTPpacket
class, which reads past the initial file header in an RTPdump
file.

Finally, a few warnings are removed.
Review URL: http://webrtc-codereview.appspot.com/138012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 08:01:16 +00:00
andrew@webrtc.org
b801571755 Comment out the ARCH_ARM define temporarily, to avoid duplicate definition conflict with the Android makefiles. 
Review URL: http://webrtc-codereview.appspot.com/135016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 18:37:59 +00:00
stefan@webrtc.org
06e2c11703 Remove unintentional printfs 
Review URL: http://webrtc-codereview.appspot.com/131018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 13:16:24 +00:00
stefan@webrtc.org
167328eab6 Disable libvpx partitions code for libvpx versions prior Cayuga. 
Necessary for WebRTC to build with Chromium. 
Also fixes the decoder wrapper's Reset() function so that properly
reinitializes the decoder.
Review URL: http://webrtc-codereview.appspot.com/132012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 13:05:48 +00:00
stefan@webrtc.org
9e812fca9f Adding missing parts related to VP8 partitions 
Review URL: http://webrtc-codereview.appspot.com/131017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 10:11:24 +00:00
stefan@webrtc.org
42ab82bf2f Disable independent partitions by default. 
Review URL: http://webrtc-codereview.appspot.com/140006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@559 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 06:55:29 +00:00
stefan@webrtc.org
c3d891059e Adds support for VP8 partitions 
This change adds support for VP8 partitions in the video jitter buffer and 
the VP8 encoder and decoder wrappers. The feature is currently disabled by
default since it requires a later version of libvpx.

With this change the jitter buffer will also start keeping track of each
packet header until decoding, and the VCMSessionInfo and VCMPacket objects 
will keep pointers into the encoded frame buffers.
Review URL: http://webrtc-codereview.appspot.com/137021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 06:50:28 +00:00