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Author SHA1 Message Date
stefan@webrtc.org
eb74a371c9 Matlab scripts useful for parsing the output from DataLog 
parseLog.m parses DataLog files.
maxUnwrap.m unwraps number sequences, useful for unwrapping e.g. 
RTP timestamp sequences and RTP sequence numbers.
Review URL: http://webrtc-codereview.appspot.com/135006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 13:24:38 +00:00
perkj@google.com
88a0da8fde Add ref_count.h to gyp file. 
Review URL: http://webrtc-codereview.appspot.com/133013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 12:51:35 +00:00
perkj@google.com
9de5917776 Add an implementation of reference count to webrtc. 
Used for instantiating objects of RefCountModule.
Review URL: http://webrtc-codereview.appspot.com/135009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 12:24:51 +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
mflodman@webrtc.org
563f658013 Adding to wathclist. 
Review URL: http://webrtc-codereview.appspot.com/139010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 07:41:05 +00:00
wu@webrtc.org
5a15ab9e36 Move the WebRtcDeviceManager and WebRtcMediaEngine to libjingle. 
Review URL: http://webrtc-codereview.appspot.com/139009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 23:04:52 +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
f1f93d822e Remove warning settings more stringent than Chromium's common.gypi. 
Review URL: http://webrtc-codereview.appspot.com/131012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:57:44 +00:00
andrew@webrtc.org
a80d026517 Fix clang warnings in voice engine. 
Review URL: http://webrtc-codereview.appspot.com/133008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:30:09 +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
andrew@webrtc.org
49e58da5b1 Fix release mode "unused variable" warnings in peerconnection. 
Review URL: http://webrtc-codereview.appspot.com/133010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:29:43 +00:00
andrew@webrtc.org
20f74285fb Temporarily switch to Chrome's hosted libvpx copy. 
Review URL: http://webrtc-codereview.appspot.com/138008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:09:16 +00:00
tommi@webrtc.org
87c546e89b Remove peerconnectionimpl_callbacks.h from libjingle.gyp. 
This file has actually never existed in trunk, but the 
line in libjingle.gyp wasn't removed when we decided not
to check in the file.  (see http://webrtc-codereview.appspot.com/60008/)
Review URL: http://webrtc-codereview.appspot.com/139011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 15:55:15 +00:00
henrik.lundin@webrtc.org
fac55d5bb7 I've added two watchlist definitions (NetEQ and video codecs), and added myself to be notified when something changes. 
Review URL: http://webrtc-codereview.appspot.com/137015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 10:29:13 +00:00
tommi@webrtc.org
c6e54a97a7 Update to the peerconnection sample app. 
* Fixes bug where remote video wasn't renderered.


* Update the Conductor class in accordance to the latest changes in the API.
  We now process the stream add/remove callbacks asynchronously.

* When a remote peer connects to us, we now call AddStream for our local streams
  to share with the peer if we haven't already done so.  To do that, we maintain
  a set of streams we have already shared.

BUG=11
Review URL: http://webrtc-codereview.appspot.com/131011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 08:37:05 +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
zakkhoyt@google.com
d9e11b429e Review URL: http://webrtc-codereview.appspot.com/137004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 00:54:32 +00:00
andrew@webrtc.org
777ef59394 Fix clang warnings in video engine. 
There are a number of namespace related warnings remaining in the video engine tests.
Review URL: http://webrtc-codereview.appspot.com/135007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 00:41:31 +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
wu@webrtc.org
b15bfd32d7 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface. 
* Fix one issue in webrtcvideoengine where we should remove the renderer before adding a new one.
Review URL: http://webrtc-codereview.appspot.com/137011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 22:14:44 +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
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