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Author SHA1 Message Date
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
mflodman@webrtc.org
cdefd423bd Adding code review watchlist to automatically CC e-mail addresses when new CLs are created. 
Review URL: http://webrtc-codereview.appspot.com/138005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 18:24:58 +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
mallinath@webrtc.org
f990eb3e88 Hi, 
Removed OnLocalStreamInitialized callback from the PeerConnection callback list. After adding OnAddStream trigger at the originator this callback was redundant. Also other modification is to provide same stream label in OnAddStream callback at the originator which provided in AddStream API.
Review URL: http://webrtc-codereview.appspot.com/138002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 17:16:35 +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
tommi@webrtc.org
8811e5af02 Switch to a smoother stretch algorithm on Windows and delete buffers from previous conversations on linux when switching back to peer list. 
Review URL: http://webrtc-codereview.appspot.com/135003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 08:39:04 +00:00
xians@google.com
3266d8d85d have the voe_cmd_test compiled with external transport enabled. 
Bug=http://code.google.com/p/webrtc/issues/detail?id=43
Test=none
Review URL: http://webrtc-codereview.appspot.com/133006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 08:29:07 +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
perkj@google.com
3fcabbe45c Modified include path after after moving files to webrtc_dev. 
Review URL: http://webrtc-codereview.appspot.com/137010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 07:44:18 +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
henrik.lundin@webrtc.org
8571af7be6 Updating to new VP8 rtp format 
The VP8 packetizer and tests have been updated to the new
RTP draft (http://tools.ietf.org/html/draft-ietf-payload-vp8-01).
The receive-side parser is also updated, and a new unit test
is implemented for it. Finally, some data traversing work to
get the parsed information into the decoder.
Review URL: http://webrtc-codereview.appspot.com/116011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 15:37:12 +00:00
hellner@google.com
09734086c6 Fixes build issue in http://code.google.com/p/webrtc/issues/detail?id=56. 
Review URL: http://webrtc-codereview.appspot.com/131008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 14:10:01 +00:00
tina.legrand@webrtc.org
81fd2bfbba New ACM codec database, created at compile time. 
Review URL: http://webrtc-codereview.appspot.com/127002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 11:18:44 +00:00
tina.legrand@webrtc.org
af931bdb39 Update of iLBC reference files for version 1.1.1, new SQRT. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 08:27:48 +00:00
tina.legrand@webrtc.org
a41b4ce7da Changing iLBC to use the new improved SQRT, WebRtcSpl_SqrtFloor(). 
The bit-stream has not change with the new SQRT, but the output signal has. The change in output is small, and all test-files pass a subjective quality test.
New test-files will be committed to svn after this CL.
Review URL: http://webrtc-codereview.appspot.com/136001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 08:19:30 +00:00
stefan@webrtc.org
c9cff24ff0 Adding classes to be used for logging data within the engines and the 
components for offline processing. Data logged with these classes can
conveniently be parsed and processed with e.g. Matlab.
Review URL: http://webrtc-codereview.appspot.com/95009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:39:02 +00:00
perkj@google.com
4094c49ddf Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC. 
Fix suggested by henrika.
Review URL: http://webrtc-codereview.appspot.com/121001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:36:28 +00:00
xians@google.com
c9b75e0a4b removing the warnings from the voe tests. 
Bug=http://code.google.com/p/webrtc/issues/detail?id=61
Test=None
Review URL: http://webrtc-codereview.appspot.com/139003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:30:16 +00:00
tina.legrand@webrtc.org
2aa5d500af Issue reported in WebRTC. A variable is defined and set, but never used. 
Review URL: http://webrtc-codereview.appspot.com/139001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 06:36:37 +00:00
henrik.lundin@webrtc.org
36450af2b3 Removing unsupported codecs from ptypes file 
The file ptypes.txt tells test program NetEqRTPplay how to
map the RTP payload types in an RTP file. Now removing payload
types that are not supported in WebRTC.
Review URL: http://webrtc-codereview.appspot.com/119009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-27 01:25:35 +00:00
mallinath@webrtc.org
92bace1faf Hi, 
This CL will support negotiation of RTCP Mux feature. Earlier we were by default enabling and assuming remote end point will support this feature as well. This will also remove the maintaining of transport channels in WebRtcSession. Its left to cricket::Transport
Review URL: http://webrtc-codereview.appspot.com/131005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-27 00:37:58 +00:00
andrew@webrtc.org
bd4494cb20 Remove the divide-by-2 when mixing. 
Review URL: http://webrtc-codereview.appspot.com/137007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 22:58:00 +00:00
mikhal@webrtc.org
b7ac56d92b video coding tests: updating quality tests following r466 
Review URL: http://webrtc-codereview.appspot.com/131009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:18:35 +00:00
mikhal@webrtc.org
d24a97fae1 video coding test: deleting unused file(resampler_test.cc) 
Review URL: http://webrtc-codereview.appspot.com/137008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:18:17 +00:00
mikhal@webrtc.org
2c3b1fb4f3 video_coding tests: removing unused functionality from test_util 
Review URL: http://webrtc-codereview.appspot.com/137009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:18:04 +00:00
mikhal@webrtc.org
a057a9561c video_coding: Updating protection logic in media optimization utility: 
1. Changing protection logic structure: Accepts only one method (not a list)
2. Removed unused code (unreferenced protection methods)
3. Removed inline constructors/destructors.  
Review URL: http://webrtc-codereview.appspot.com/120005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:17:34 +00:00
mikhal@webrtc.org
552f173979 video_coding: Moving video metrics computation to a designated file. 
This is the first stage of a general clean-up to test_util. Will try to divide this clean-up to small changes, so it will be easier to review. 
Review URL: http://webrtc-codereview.appspot.com/120004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:38:09 +00:00
andrew@webrtc.org
e46d69f762 Fix gcc 4.6 set but unused warnings in AEC. 
Review URL: http://webrtc-codereview.appspot.com/134003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:20:54 +00:00
mallinath@webrtc.org
b62c776eca moving all new version related files to webrtc_dev and removed from webrtc. 
Review URL: http://webrtc-codereview.appspot.com/138001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:19:09 +00:00
andrew@webrtc.org
ffbe7a75fd Cast away the unused state argument value to silence gcc 4.6 warnings. 
The WebRTC C wrapper for the G711 codec doesn't actually use the 'state' 
argument, but declares one anyway for API uniformity.

At the beginning of functions like WebRTCG711_EncodeA(), there's a stanza:

    // Set to avoid getting warnings
    state = NULL;

This might work around an unused parameter warning, but under gcc 4.6.0 
it ends up generating another warning, that state is set but not used.  

Casting the assignment to void silences the warning, restoring 
compilation under -Werror.

Reported as https://code.google.com/p/webrtc/issues/detail?id=50
Review URL: http://webrtc-codereview.appspot.com/135002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:16:30 +00:00
turajs@google.com
7f2bbbbefd To remove all calls involving scratch-memory 
Review URL: http://webrtc-codereview.appspot.com/129001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 16:03:49 +00:00
turajs@google.com
ac55f7b33c Review URL: http://webrtc-codereview.appspot.com/115004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 16:02:16 +00:00
xians@google.com
7659b366ac revert the file path in the voe_auto_test 
Review URL: http://webrtc-codereview.appspot.com/131007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 14:13:27 +00:00
tommi@webrtc.org
350d091e0e Send the hangup message when asked to disconnect from a peer. 
Review URL: http://webrtc-codereview.appspot.com/131006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 13:20:41 +00:00
xians@webrtc.org
c57f9c38ad Using IAudioEndpointVolume in IsSpeakerMuteAvailable and IsMicrophoneMuteAvailable to be consistent with SpeakerMute and MicrophoneMute APIs. 
Review URL: http://webrtc-codereview.appspot.com/112007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 12:28:33 +00:00
mflodman@webrtc.org
4fcb0caf78 Removing warning in video capture module for linux and auto test. 
Review URL: http://webrtc-codereview.appspot.com/134002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 10:54:48 +00:00
hellner@google.com
b55c988b22 Updated peerconnection_unittest slightly. Also added it to the build. 
Review URL: http://webrtc-codereview.appspot.com/133003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 23:01:40 +00:00
hellner@google.com
23a8065e36 Fixed broken build due to r453. 
Review URL: http://webrtc-codereview.appspot.com/131004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 21:40:11 +00:00
hellner@google.com
b2801f3a16 Added the remaining test cases for the webrtcsession unittest also some minor refactoring. 
Review URL: http://webrtc-codereview.appspot.com/131003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 21:37:08 +00:00
zakkhoyt@google.com
59af6f1434 Porting Mac keypress detection from GIPS repository. 
Mac keypress detection was added specifically for GTalk.
Review URL: http://webrtc-codereview.appspot.com/124001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 20:30:25 +00:00
mikhal@webrtc.org
ba9bd692ea video_coding_tests: Fix build error 
Review URL: http://webrtc-codereview.appspot.com/132001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 20:12:03 +00:00
andrew@webrtc.org
aed0348e5b Roll gyp 985:1012 
Fix the world rebuilding in make 3.82.
http://code.google.com/p/webrtc/issues/detail?id=62

r1012 also allows Chrome to build with Make on Mac. Haven't tested WebRTC, but it would be nice to have.
Review URL: http://webrtc-codereview.appspot.com/119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 18:45:51 +00:00