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Author SHA1 Message Date
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
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
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
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
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
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
henrik.lundin@webrtc.org
dd07d5932a Let VP8 decoder handle NULL codecSpecificInfo 
VP8Decoder::Decode() can now handle the case when
codecSpecificInfo is NULL. Previously, it would crash.

Review URL: http://webrtc-codereview.appspot.com/135015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:21:38 +00:00
henrik.lundin@webrtc.org
ea05973e68 Fixing VCM tests for VP8 
Removing asserts since the PictureID (and other parameters)
is now piped through codecSpecific. Also made sure the VCM
send callbacks (test code) copies the appropriate paramters.
Finally, enabling I420 in tests.

Review URL: http://webrtc-codereview.appspot.com/137022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:20:17 +00:00
henrika@google.com
73d65513f1 Adds reference counting to the ADM. 
This CL modifies the ADM interface to ensure that an external ADM
can't call Create and Destroy any longer.

It also contains some minor style nits to conform better with
the Chromium style guide.
Review URL: http://webrtc-codereview.appspot.com/133014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:11:18 +00:00
andrew@webrtc.org
b44172dab9 Fix "braces recommended" warning in audio_conference_mixer. 
Review URL: http://webrtc-codereview.appspot.com/131014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 18:04:32 +00:00
perkj@google.com
ac75cab618 Fix reference counting assert. 
Change assert("teo") to assert(!"teo") so that the assert is actually triggered.
Review URL: http://webrtc-codereview.appspot.com/133018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 13:58:34 +00:00
stefan@webrtc.org
269f8a14c6 Undoing change committed in r514 since it broke bandwidth estimation 
Review URL: http://webrtc-codereview.appspot.com/132011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@531 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 09:51:59 +00:00
perkj@google.com
ea72c34fb9 Temporary add dummy implementation to RefCountModule. The reason is so that ADM and VideoCapture implementations can change to refcounted versions before forcing them. 
Review URL: http://webrtc-codereview.appspot.com/139014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-05 11:11:04 +00:00
henrik.lundin@webrtc.org
1e53166569 Fix VP8 tests 
These are changes that make the VP8 tests work again after the
wrapper was updated. The codec specific info is now propagated
properly through the encoder callback and into the queue struct.

Also added an fclose to get rid of a valgrind warning.
Review URL: http://webrtc-codereview.appspot.com/138011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@526 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-05 07:23:33 +00:00
zakkhoyt@google.com
fb298d3783 Modified path on speex lib 
Review URL: http://webrtc-codereview.appspot.com/137018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 22:06:49 +00:00
andrew@webrtc.org
413b993166 Put some table size information in one place. 
Motivated by fixing an unused variable warning in release mode.
Review URL: http://webrtc-codereview.appspot.com/132007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 22:03:56 +00:00
turajs@google.com
d7a41774ce header included twice. 
Review URL: http://webrtc-codereview.appspot.com/139013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 20:52:47 +00:00
henrik.lundin@webrtc.org
2641fd1d19 Remove warnings in vp8_test 
Most modifications are either reordering of the initializers in constructors, removed unused variables, or comparison mismatches taken care of. A few other special cases are commented.
Review URL: http://webrtc-codereview.appspot.com/132008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 12:09:07 +00:00
perkj@google.com
ef04cf4b2e Adding reference counted version of the module interface. 
The reason for this is that we would like to have reference counting on the modules you can register externally with ViE and VoE.
Currently we plan to use this on the ADM, VideoCapture module and VideoRenderModule.
Review URL: http://webrtc-codereview.appspot.com/138010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 09:47:28 +00:00
andrew@webrtc.org
4d905f88c6 Fix clang warnings in rtp. 
Review URL: http://webrtc-codereview.appspot.com/132006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 19:22:27 +00:00
andrew@webrtc.org
bbd8908664 Fix clang warnings in video coding. 
Review URL: http://webrtc-codereview.appspot.com/138007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:30:01 +00:00
tina.legrand@webrtc.org
84519ec0a2 Fixing some inconsistencies in WebRTC audio coding module. I've added setup information for all codecs which are not part of WebRTC, but possible to hook in. 
Please help me review.
Henrik: review neteq_defines.h
Turaj: review all files, but the one Henrik reviews.
Zakk: FYI only.
Review URL: http://webrtc-codereview.appspot.com/138004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 07:47:31 +00:00
marpan@google.com
243db12616 media_opt_util: Fixed an assert and some code cleanup for AvgRecoveryFEC function. 
Review URL: http://webrtc-codereview.appspot.com/139007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 22:14:52 +00:00
turajs@google.com
ebb2744337 To fix warning for unused variable. And fix some warning in test. 
Review URL: http://webrtc-codereview.appspot.com/131010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 21:28:08 +00:00
turajs@google.com
eaf3185105 Take care of unused variable. 
Review URL: http://webrtc-codereview.appspot.com/137013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 21:27:53 +00:00
andrew@webrtc.org
9562a3664c Last fixes to build with gcc 4.6. 
Set but unused parameter/variable warnings.
http://code.google.com/p/webrtc/issues/detail?id=52
Review URL: http://webrtc-codereview.appspot.com/139006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 18:50:12 +00:00
andrew@webrtc.org
830099eba4 Add a gyp flag to disable video functionality from dependencies shared by voice and video engine. 
Currently, this is just the utility module. It relies on the already existing WEBRTC_MODULE_UTILITY_VIDEO define.
Review URL: http://webrtc-codereview.appspot.com/133007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 17:03:54 +00:00
pwestin@webrtc.org
e9f0e2eb20 Moved _rtpReceiver to protected 
Review URL: http://webrtc-codereview.appspot.com/132005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 13:16:52 +00:00
tommi@webrtc.org
c7d5f6249b Fix build errors on Windows. 
Since this is a C file, variables must be declared at the top of the function
so I'm moving the fix for the warning (inst = NULL) to the bottom of the funciton.
Otherwise, the compiler will complain when it sees int i; on systems that do
not have WEBRTC_BIG_ENDIAN defined.
Review URL: http://webrtc-codereview.appspot.com/139005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 12:11:24 +00:00
turajs@google.com
74c640aebb fix build break 
Review URL: http://webrtc-codereview.appspot.com/132004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 20:44:24 +00:00
turajs@google.com
7796c02b42 Wrap encode, decode, PLC NB functions in #define to avoid warnings. 
Review URL: http://webrtc-codereview.appspot.com/133005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 20:30:17 +00:00
turajs@google.com
8ecd0e8f3d Remove Clang warning for PCM16B. 
Review URL: http://webrtc-codereview.appspot.com/137006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 20:29:50 +00:00
punyabrata@google.com
eba8c32840 Resolving a race condition issue related to using shared devices 
(e.g. usb headsets) where we were not stopped the shared callback
until both StopPlayout() and StopRecording() are called. Google
internal bugid 4478351
Review URL: http://webrtc-codereview.appspot.com/130001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 14:32:22 +00:00
xians@google.com
e74a9ea303 AudioDeviceUtility::WaitForKey() pulls two characters if the first one is a newline, but discards the final value. 
The current code assigns that second value to a local variable, which generates a set-but-unused warning on gcc 4.6.0. Instead, cast the result away.

I also refactor the code a bit by adding the right indentation and removing empty lines.

Bug=http://code.google.com/p/webrtc/issues/detail?id=53
Test=none
Review URL: http://webrtc-codereview.appspot.com/135005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 08:27:02 +00:00
xians@google.com
932096c84f Porting gtalk alsa impl from depot to webrtc 
Review URL: http://webrtc-codereview.appspot.com/123002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 07:41:55 +00:00
mikhal@webrtc.org
46171cf546 video coding tests: Adding a Normal distribution to simulate packet arrival times 
Review URL: http://webrtc-codereview.appspot.com/138003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 23:38:04 +00:00