mflodman@webrtc.org
cdefd423bd
Adding code review watchlist to automatically CC e-mail addresses when new CLs are created.
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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.
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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
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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.
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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
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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.
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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.
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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,
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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
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(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.
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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.
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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.
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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.
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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
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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
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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
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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 .
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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.
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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.
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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().
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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.
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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.
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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.
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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.
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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
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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)
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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
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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:
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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.
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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.
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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.
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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
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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
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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
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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
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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
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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
hellner@google.com
40373cc184
Bugfix in unittest and some minor refactoring.
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Review URL: http://webrtc-codereview.appspot.com/137003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@450 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-25 17:17:30 +00:00
wu@webrtc.org
eb9572e501
Add the new peerconnection factory to the scons file.
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Review URL: http://webrtc-codereview.appspot.com/134001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@449 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-25 16:58:58 +00:00
niklas.enbom@webrtc.org
e129ae944e
Review URL: http://webrtc-codereview.appspot.com/137002
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@448 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-25 16:52:34 +00:00