external/webrtc.git
32 hours agoMove Jingle-specific files from talk/session/media to webrtc/libjingle/session/media... master
pthatcher@webrtc.org [Fri, 19 Dec 2014 22:29:55 +0000 (22:29 +0000)]
Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media.  This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository.

Also, fix the includes and header guards of examples/call.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7972 4adac7df-926f-26a2-2b94-8c16560cd09d

33 hours agoFix mac video capture leak.
tkchin@webrtc.org [Fri, 19 Dec 2014 20:51:02 +0000 (20:51 +0000)]
Fix mac video capture leak.

BUG=3878
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7971 4adac7df-926f-26a2-2b94-8c16560cd09d

34 hours agoAdd initWithCoder to RTCEAGLVideoView.
tkchin@webrtc.org [Fri, 19 Dec 2014 20:47:35 +0000 (20:47 +0000)]
Add initWithCoder to RTCEAGLVideoView.

Allows for proper OpenGL initialization if view is created from
storyboard.

BUG=3896
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7970 4adac7df-926f-26a2-2b94-8c16560cd09d

34 hours agoWire up Beamformer in AudioProcessing
aluebs@webrtc.org [Fri, 19 Dec 2014 19:57:34 +0000 (19:57 +0000)]
Wire up Beamformer in AudioProcessing

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7969 4adac7df-926f-26a2-2b94-8c16560cd09d

36 hours agoFix the ramp-up-down-up test which was using ts-offset extension with the abs-send...
stefan@webrtc.org [Fri, 19 Dec 2014 18:00:21 +0000 (18:00 +0000)]
Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator.

BUG=chromium:444023
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7968 4adac7df-926f-26a2-2b94-8c16560cd09d

36 hours agoRemove unneccessary lock causing a potential deadlock.
stefan@webrtc.org [Fri, 19 Dec 2014 17:55:20 +0000 (17:55 +0000)]
Remove unneccessary lock causing a potential deadlock.

TBR=pbos@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7967 4adac7df-926f-26a2-2b94-8c16560cd09d

37 hours agoAdd a AppRTCDemo setting to change the GAE server.
jiayl@webrtc.org [Fri, 19 Dec 2014 17:32:14 +0000 (17:32 +0000)]
Add a AppRTCDemo setting to change the GAE server.

BUG=4041
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7966 4adac7df-926f-26a2-2b94-8c16560cd09d

39 hours agoRemove the last getters from VideoReceiveStream stats.
pbos@webrtc.org [Fri, 19 Dec 2014 15:45:03 +0000 (15:45 +0000)]
Remove the last getters from VideoReceiveStream stats.

R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/32899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d

39 hours agoEnable payload-based padding by default and remove the API.
stefan@webrtc.org [Fri, 19 Dec 2014 15:33:17 +0000 (15:33 +0000)]
Enable payload-based padding by default and remove the API.

BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d

40 hours agoUnify the two copies of move.h
kwiberg@webrtc.org [Fri, 19 Dec 2014 14:35:57 +0000 (14:35 +0000)]
Unify the two copies of move.h

This patch basically deletes webrtc/base/move.h (which is the more
outdated copy) and moves webrtc/system_wrappers/source/move.h to take
its place.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7963 4adac7df-926f-26a2-2b94-8c16560cd09d

40 hours agoRtp-Rtcp sender cleanup.
pbos@webrtc.org [Fri, 19 Dec 2014 13:49:55 +0000 (13:49 +0000)]
Rtp-Rtcp sender cleanup.

Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.

Also removed const on non-pointer/reference types for related files.

BUG=
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34469004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d

41 hours agoGN: Fix build for Mac
kjellander@webrtc.org [Fri, 19 Dec 2014 13:28:37 +0000 (13:28 +0000)]
GN: Fix build for Mac

BUG=4105
R=henrika@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7961 4adac7df-926f-26a2-2b94-8c16560cd09d

44 hours agoMove updating nack bitrate inside UpdateNACKBitRate.
stefan@webrtc.org [Fri, 19 Dec 2014 09:52:24 +0000 (09:52 +0000)]
Move updating nack bitrate inside UpdateNACKBitRate.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoBreakup Transports and TransportParsers and move TransportParsers into webrtc/libjing...
pthatcher@webrtc.org [Fri, 19 Dec 2014 03:32:59 +0000 (03:32 +0000)]
Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle.  This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoMerge beamformer
aluebs@webrtc.org [Thu, 18 Dec 2014 22:22:04 +0000 (22:22 +0000)]
Merge beamformer

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7958 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRemove obsolete target_arch == armv7.
andrew@webrtc.org [Thu, 18 Dec 2014 21:36:18 +0000 (21:36 +0000)]
Remove obsolete target_arch == armv7.

Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.

BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com

Review URL: https://webrtc-codereview.appspot.com/38379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoSplit up (Jingle)Session from BaseSession. This is part of an ongoing effort to...
pthatcher@webrtc.org [Thu, 18 Dec 2014 20:31:29 +0000 (20:31 +0000)]
Split up (Jingle)Session from BaseSession.  This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.

(This is the 3rd try)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoClean up the Channel code in AppRTCDemo and use GAE prod server for new signaling...
jiayl@webrtc.org [Thu, 18 Dec 2014 20:12:03 +0000 (20:12 +0000)]
Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7955 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoMove session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move...
pthatcher@webrtc.org [Thu, 18 Dec 2014 17:09:11 +0000 (17:09 +0000)]
Move session/tunnel to webrtc/libjingle.  This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository.   I won't submit this until all other projects have moved off of compiling this as well.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7953 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoStore the received report blocks map (mapped per remote ssrc) in a map per source...
asapersson@webrtc.org [Thu, 18 Dec 2014 14:30:32 +0000 (14:30 +0000)]
Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc.
When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.

Removed unused function ResetRTT.

BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33659005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRefactor some receive-side stats.
pbos@webrtc.org [Thu, 18 Dec 2014 13:50:16 +0000 (13:50 +0000)]
Refactor some receive-side stats.

Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoGet avg_delay_ms from DecoderTiming callback.
pbos@webrtc.org [Thu, 18 Dec 2014 13:12:52 +0000 (13:12 +0000)]
Get avg_delay_ms from DecoderTiming callback.

R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7949 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoSuppress REMB in bitrate ctrl if it seems lika a short network glitch.
sprang@webrtc.org [Thu, 18 Dec 2014 11:53:59 +0000 (11:53 +0000)]
Suppress REMB in bitrate ctrl if it seems lika a short network glitch.

BUG=4082
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7948 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRemove _t from function pointer typedefs.
pbos@webrtc.org [Thu, 18 Dec 2014 09:56:09 +0000 (09:56 +0000)]
Remove _t from function pointer typedefs.

_t are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7947 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoMake an AudioEncoder subclass for iSAC redundant encoding
henrik.lundin@webrtc.org [Thu, 18 Dec 2014 09:52:36 +0000 (09:52 +0000)]
Make an AudioEncoder subclass for iSAC redundant encoding

Adding unit test, too.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7946 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRename rtpDumpPktHdr_t to RtpDumpPacketHeader.
pbos@webrtc.org [Thu, 18 Dec 2014 09:18:42 +0000 (09:18 +0000)]
Rename rtpDumpPktHdr_t to RtpDumpPacketHeader.

_t names are reserved in POSIX.

BUG=162
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7945 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRename external_hmac_ctx_t to ExternalHmacContext.
pbos@webrtc.org [Thu, 18 Dec 2014 09:12:21 +0000 (09:12 +0000)]
Rename external_hmac_ctx_t to ExternalHmacContext.

_t types are reserved by POSIX.

R=juberti@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/33699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7944 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRename _t struct types in audio_processing.
pbos@webrtc.org [Thu, 18 Dec 2014 09:11:33 +0000 (09:11 +0000)]
Rename _t struct types in audio_processing.

_t names are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7943 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoFixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder
henrik.lundin@webrtc.org [Thu, 18 Dec 2014 06:58:42 +0000 (06:58 +0000)]
Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder

Re-enable the test and explicitly call delete on red, even though the
test should die in the AudioEncoderCopyRed cunstructor. Apparently,
things work a little differently under memcheck.

BUG=4108, 3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7942 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoWorkaround for issue 3927 to allow localhost IP even if it doesn't match the local...
guoweis@webrtc.org [Thu, 18 Dec 2014 04:45:05 +0000 (04:45 +0000)]
Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port

BUG=3927
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7941 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRevert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort...
pthatcher@webrtc.org [Thu, 18 Dec 2014 02:28:25 +0000 (02:28 +0000)]
Revert "Split up (Jingle)Session from BaseSession.  This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."

This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.

BUG=
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoSplit up (Jingle)Session from BaseSession. This is part of an ongoing effort to...
pthatcher@webrtc.org [Thu, 18 Dec 2014 01:22:02 +0000 (01:22 +0000)]
Split up (Jingle)Session from BaseSession.  This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.

This is an un-revert of r7992 and r7993.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoFix an assert failure caused by race condition
guoweis@webrtc.org [Thu, 18 Dec 2014 00:30:55 +0000 (00:30 +0000)]
Fix an assert failure caused by race condition

BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7938 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoMake safe_conversions suitable for rtc_base_approved.
andrew@webrtc.org [Wed, 17 Dec 2014 22:56:09 +0000 (22:56 +0000)]
Make safe_conversions suitable for rtc_base_approved.

Since we want to use checked_cast in WavReader.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7937 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoMove jingle examples from talk/ into webrtc/libjingle. This is part of the effor...
pthatcher@webrtc.org [Wed, 17 Dec 2014 22:15:11 +0000 (22:15 +0000)]
Move jingle examples from talk/ into webrtc/libjingle.  This is part of the effor to move Jingle out of WebRTC and into its own repository.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7936 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoMove VirtualSocket into the .h file to allow unit tests more control over behavior.
guoweis@webrtc.org [Wed, 17 Dec 2014 22:03:33 +0000 (22:03 +0000)]
Move VirtualSocket into the .h file to allow unit tests more control over behavior.

BUG=3927
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7935 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoSupport block_size greater than chunk_size in Blocker
aluebs@webrtc.org [Wed, 17 Dec 2014 17:28:31 +0000 (17:28 +0000)]
Support block_size greater than chunk_size in Blocker

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7934 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRename _t struct types in audio_coding.
pbos@webrtc.org [Wed, 17 Dec 2014 15:23:29 +0000 (15:23 +0000)]
Rename _t struct types in audio_coding.

_t names are reserved in POSIX.

R=henrik.lundin@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoChange MockStatsObserver to grab values inside of OnComplete.
tommi@webrtc.org [Wed, 17 Dec 2014 14:09:05 +0000 (14:09 +0000)]
Change MockStatsObserver to grab values inside of OnComplete.
This is done since StatsReportCopyable is going away and the list of
supported properties of the mock class is known.
StatsReports holds a list of pointers to objects that cannot be cached,
so this is a simple way to grab the values when they're available.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7932 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRemove or rename typedefs with _t prefixes.
pbos@webrtc.org [Wed, 17 Dec 2014 13:43:55 +0000 (13:43 +0000)]
Remove or rename typedefs with _t prefixes.

_t prefixes are reserved for additional typenames in POSIX.

R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/36559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoAdd a little utility to capture cpu graphs.
tommi@webrtc.org [Wed, 17 Dec 2014 12:35:29 +0000 (12:35 +0000)]
Add a little utility to capture cpu graphs.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7930 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoAdd overshoot of target bitrate for screenshare with temporal layers.
sprang@webrtc.org [Wed, 17 Dec 2014 10:57:10 +0000 (10:57 +0000)]
Add overshoot of target bitrate for screenshare with temporal layers.

Set the codec target bitrate higher than TL0 but lower than TL1, making
sure frame rate is not too low (but still lower than TL1) and that
overshooting for complex scenes don't overly exceed TL1 bitrates.

BUG=4083
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7929 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoChange aggregated fraction loss to be calculated from the cumulative loss and extende...
asapersson@webrtc.org [Wed, 17 Dec 2014 10:27:57 +0000 (10:27 +0000)]
Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls.

Previously it was derived from the fraction loss of the current report (which could be based on a received report block in between two get stats calls).

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7928 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoEnable the iSACfix AudioDecoder test (and make it work again)
kwiberg@webrtc.org [Wed, 17 Dec 2014 07:30:23 +0000 (07:30 +0000)]
Enable the iSACfix AudioDecoder test (and make it work again)

As far as I can tell, the test should have been enabled again once
https://code.google.com/p/webrtc/issues/detail?id=1353 was fixed, but
it wasn't, and has rotted a bit as a result. I'm not sure why the
number of encoded bytes have changed, but the output seems to be
correct (EncodeDecodeTest encodes, decodes, and compares the result
with the original).

The DecodePlc change is necessary because r7912 added support for that
to the iSACfix AudioDecoder.

BUG=1353, 3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7927 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoIf one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSup...
braveyao@webrtc.org [Wed, 17 Dec 2014 05:59:41 +0000 (05:59 +0000)]
If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport().
Verified in chromium. Now the existing content still could work.

BUG=4096
TEST=Manual Test
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7926 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd adapter_type into Candidate object.
guoweis@webrtc.org [Tue, 16 Dec 2014 23:01:31 +0000 (23:01 +0000)]
Add adapter_type into Candidate object.

Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Committed: https://code.google.com/p/webrtc/source/detail?r=7906

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix path to mock_agc.h
andrew@webrtc.org [Tue, 16 Dec 2014 22:28:20 +0000 (22:28 +0000)]
Fix path to mock_agc.h

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7924 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoRevert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort...
pthatcher@webrtc.org [Tue, 16 Dec 2014 22:28:03 +0000 (22:28 +0000)]
Revert "Split up (Jingle)Session from BaseSession.  This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."

This reverts r7992.

It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoSplit up (Jingle)Session from BaseSession. This is part of an ongoing effort to...
pthatcher@webrtc.org [Tue, 16 Dec 2014 21:37:37 +0000 (21:37 +0000)]
Split up (Jingle)Session from BaseSession.  This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMove ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediase...
pthatcher@webrtc.org [Tue, 16 Dec 2014 21:09:08 +0000 (21:09 +0000)]
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h.  This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoDisable AudioEncoderCopyRedDeathTest.NullSpeechEncoder
henrik.lundin@webrtc.org [Tue, 16 Dec 2014 21:04:55 +0000 (21:04 +0000)]
Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder

Fails linux memcheck.

BUG=4108
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7920 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMake one OWNERS files for all of webrtc/libjingle so we don't need approval from...
pthatcher@webrtc.org [Tue, 16 Dec 2014 21:04:41 +0000 (21:04 +0000)]
Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7919 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd a manageable command-line tool for AudioProcessing.
andrew@webrtc.org [Tue, 16 Dec 2014 20:57:15 +0000 (20:57 +0000)]
Add a manageable command-line tool for AudioProcessing.

This is the start of a replacement for the venerable and unwieldly
process_test.cc (aka audioproc). It will be limited to:
- Reading WAV or aecdebug protobuf files.
- Calling the float AudioProcessing interface.
- Requiring aecdebug files for running bi-directional stream
components (e.g. AEC).

This initial version only handles WAV files.

R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7918 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd 48kHz support to AGC
aluebs@webrtc.org [Tue, 16 Dec 2014 20:56:09 +0000 (20:56 +0000)]
Add 48kHz support to AGC

Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

Originally reviewed here:
https://webrtc-codereview.appspot.com/26339004/

BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7917 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd (safe) uint32_t cast to fix Win64 build.
andrew@webrtc.org [Tue, 16 Dec 2014 20:47:42 +0000 (20:47 +0000)]
Add (safe) uint32_t cast to fix Win64 build.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7916 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoHandle all permissible PCM fields with WavReader.
andrew@webrtc.org [Tue, 16 Dec 2014 20:17:21 +0000 (20:17 +0000)]
Handle all permissible PCM fields with WavReader.

I discovered the hard way that Adobe Audition writes an 18 byte format
header with an extra (zero) extension size field. Although:
https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
indicates this field shouldn't exist for PCM, the documentation here:
http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/WAVE/WAVE.html
doesn't list it as strictly forbidden, only that it _must_ exist for
non-PCM formats.

Audition can write metadata to the file after the audio data, which is
also not forbidden. We now ensure to read only up to the audio payload
length to avoid reading the metadata.

R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7915 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd AGC manager tests.
pbos@webrtc.org [Tue, 16 Dec 2014 14:48:47 +0000 (14:48 +0000)]
Add AGC manager tests.

R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/35539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7914 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMake an AudioEncoder subclass for RED
henrik.lundin@webrtc.org [Tue, 16 Dec 2014 13:41:36 +0000 (13:41 +0000)]
Make an AudioEncoder subclass for RED

This class only supports the simple case of payload duplication. That
is, one single encoder is used, and the redundant payload is a one-step
delayed payload.

BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAudioEncoder subclass for iSACfix
kwiberg@webrtc.org [Tue, 16 Dec 2014 12:49:37 +0000 (12:49 +0000)]
AudioEncoder subclass for iSACfix

This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoCleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().
kjellander@webrtc.org [Tue, 16 Dec 2014 12:29:59 +0000 (12:29 +0000)]
Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().

This should fix the following error I'm seeing in Win8 GN trybot:

e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(78)
: error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(30)
: warning C4373:
'webrtc::BitrateControllerImpl::RtcpBandwidthObserverImpl::OnReceivedEstimatedBitrate':
virtual function overrides 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate',
previous versions of the compiler did not override when parameters only differed by const/volatile qualifiers
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\rtp_rtcp\interface\rtp_rtcp_defines.h(286)
: see declaration of 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate'

http://build.chromium.org/p/tryserver.chromium.win/builders/win8_chromium_gn_dbg/builds/23/steps/compile/logs/stdio

The above was triggered in CL https://codereview.chromium.org/802113002/

BUG=None
R=kjellander@google.com, kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37409004

Patch from Thiago Farina <tfarina@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7911 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd field to counters for when first rtp/rtcp packet is sent/received.
asapersson@webrtc.org [Tue, 16 Dec 2014 12:03:11 +0000 (12:03 +0000)]
Add field to counters for when first rtp/rtcp packet is sent/received.
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoaudio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/
bjornv@webrtc.org [Tue, 16 Dec 2014 10:38:10 +0000 (10:38 +0000)]
audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/

include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.

BUG=
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRevert "Add adapter_type into Candidate object."
guoweis@webrtc.org [Tue, 16 Dec 2014 05:28:10 +0000 (05:28 +0000)]
Revert "Add adapter_type into Candidate object."

This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.

BUG=
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoFix vp9 setting in vie loopback test.
marpan@webrtc.org [Tue, 16 Dec 2014 00:21:47 +0000 (00:21 +0000)]
Fix vp9 setting in vie loopback test.

If vp9 codec was selected then videoCodec.codecSpecific.VP8.numberOfTemporalLayers was being set.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/37389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7907 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd adapter_type into Candidate object.
guoweis@webrtc.org [Mon, 15 Dec 2014 23:03:10 +0000 (23:03 +0000)]
Add adapter_type into Candidate object.

Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoUse int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
pkasting@chromium.org [Mon, 15 Dec 2014 22:09:40 +0000 (22:09 +0000)]
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemove 20ms support in AGC
aluebs@webrtc.org [Mon, 15 Dec 2014 21:54:50 +0000 (21:54 +0000)]
Remove 20ms support in AGC

Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoReenable test case P2PTransportChannelTest.TestIPv6Connections
guoweis@webrtc.org [Mon, 15 Dec 2014 21:25:54 +0000 (21:25 +0000)]
Reenable test case P2PTransportChannelTest.TestIPv6Connections

BUG=3317
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7903 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoMerge in AGC manager and AGC tools.
pbos@webrtc.org [Mon, 15 Dec 2014 16:33:16 +0000 (16:33 +0000)]
Merge in AGC manager and AGC tools.

R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/37379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemoves unused test files by audio_processing/transient
bjornv@webrtc.org [Mon, 15 Dec 2014 16:13:05 +0000 (16:13 +0000)]
Removes unused test files by audio_processing/transient

BUG=
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7901 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoresources/audio_processing: Removed unused test files
bjornv@webrtc.org [Mon, 15 Dec 2014 15:57:11 +0000 (15:57 +0000)]
resources/audio_processing: Removed unused test files

Two files not used by any tests are removed.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7900 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoSuppressing warnings in GetRTT() in VoE.
minyue@webrtc.org [Mon, 15 Dec 2014 14:56:44 +0000 (14:56 +0000)]
Suppressing warnings in GetRTT() in VoE.

GetRTT() was separated from GetRTPStatistics() but the warnings were not updated.

Now GetRTT() is only only used by GetRTPStatistics() and the warning pops up pointlessly and too often.

This CL is to suppress these warnings and maintain a proper warning for GetRTPStatistics().

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7899 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoClean up StatsObserver's OnComplete methods (address TODOs).
tommi@webrtc.org [Mon, 15 Dec 2014 13:22:54 +0000 (13:22 +0000)]
Clean up StatsObserver's OnComplete methods (address TODOs).

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoUse webrtc_root instead of DEPTH for iSAC.
pbos@webrtc.org [Mon, 15 Dec 2014 10:56:50 +0000 (10:56 +0000)]
Use webrtc_root instead of DEPTH for iSAC.

Un-breaks chromium.webrtc.fyi. Broken as Chromium doesn't have webrtc/
checked out in root.

TBR=bjornv@webrtc.org,tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7897 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days ago(Auto)update libjingle 82121498-> 82126219
buildbot@webrtc.org [Mon, 15 Dec 2014 09:48:07 +0000 (09:48 +0000)]
(Auto)update libjingle 82121498-> 82126219

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemove unneeded ctor and add a more practical one
tommi@webrtc.org [Mon, 15 Dec 2014 09:47:49 +0000 (09:47 +0000)]
Remove unneeded ctor and add a more practical one
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd thread asserts to StatsCollector.
tommi@webrtc.org [Mon, 15 Dec 2014 09:44:48 +0000 (09:44 +0000)]
Add thread asserts to StatsCollector.
Also adding a "ForTest" postfix to a test-only method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoMerge audio_processing changes.
pbos@webrtc.org [Mon, 15 Dec 2014 09:41:24 +0000 (09:41 +0000)]
Merge audio_processing changes.

R=aluebs@webrtc.org, bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRevert r7885.
pbos@webrtc.org [Mon, 15 Dec 2014 08:04:50 +0000 (08:04 +0000)]
Revert r7885.

Breaks compile step of other code where network name of
cricket::Candidate is used.

TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
andrew@webrtc.org [Mon, 15 Dec 2014 07:23:49 +0000 (07:23 +0000)]
Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.

This intrinsics version gives bit-exact result as the current assembly
neon code. And the performance is 38% better than current assembly
neon version, 5.92 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I257e33ef6d634a519fd71adc4f52b06dd655bd9d

Review URL: https://webrtc-codereview.appspot.com/32749004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7891 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRevert r7886:7887.
pbos@webrtc.org [Mon, 15 Dec 2014 07:03:04 +0000 (07:03 +0000)]
Revert r7886:7887.

Broke build steps in other code that uses securetunnelsessionclient.cc
and others.

TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoAdd NEON intrinsics version for min_max_operations_neon.c
andrew@webrtc.org [Mon, 15 Dec 2014 06:07:47 +0000 (06:07 +0000)]
Add NEON intrinsics version for min_max_operations_neon.c

WebRtcSpl_MinValueW32Neon, WebRtcSpl_MaxValueW32Neon, WebRtcSpl_MaxValueW16Neon
and WebRtcSpl_MaxAbsValueW32Neon are added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id461d64c3313f56147edadd2231e8845574ead2a

Review URL: https://webrtc-codereview.appspot.com/28259004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7889 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoMove WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
magjed@webrtc.org [Sun, 14 Dec 2014 11:09:23 +0000 (11:09 +0000)]
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h

The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoPut pseudotcp back because remoting uses it.
pthatcher@webrtc.org [Sat, 13 Dec 2014 01:56:39 +0000 (01:56 +0000)]
Put pseudotcp back because remoting uses it.

R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7887 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoMove the obvious/easy Jingle-specific code into webrtc/libjingle.
pthatcher@webrtc.org [Fri, 12 Dec 2014 21:04:42 +0000 (21:04 +0000)]
Move the obvious/easy Jingle-specific code into webrtc/libjingle.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdd adapter_type into Candidate object.
guoweis@webrtc.org [Fri, 12 Dec 2014 19:21:14 +0000 (19:21 +0000)]
Add adapter_type into Candidate object.

Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoSwitch kStatsValueName* constants to be enums instead of char*.
tommi@webrtc.org [Fri, 12 Dec 2014 17:41:28 +0000 (17:41 +0000)]
Switch kStatsValueName* constants to be enums instead of char*.
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoMoving encoded_bytes into EncodedInfo
henrik.lundin@webrtc.org [Fri, 12 Dec 2014 13:31:24 +0000 (13:31 +0000)]
Moving encoded_bytes into EncodedInfo

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoFix webrtc gn windows build.
kjellander@webrtc.org [Fri, 12 Dec 2014 12:10:46 +0000 (12:10 +0000)]
Fix webrtc gn windows build.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7882 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoRemoving manual test pages because they have been moved to github.
jansson@webrtc.org [Fri, 12 Dec 2014 09:30:41 +0000 (09:30 +0000)]
Removing manual test pages because they have been moved to github.

BUG=https://github.com/GoogleChrome/webrtc/issues/203
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7881 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoCleanup little things found when refactoring.
pthatcher@webrtc.org [Fri, 12 Dec 2014 02:44:30 +0000 (02:44 +0000)]
Cleanup little things found when refactoring.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMove the downmixing out of AudioBuffer
aluebs@webrtc.org [Thu, 11 Dec 2014 17:09:21 +0000 (17:09 +0000)]
Move the downmixing out of AudioBuffer

This provides more flexibility if some component in AudioProcessing wants to operate before downmixing.
Now the AudioProcessing does only track the processing rate, but not the processing number of channels. This is tracked by the AudioBuffer itself and can be changed at any time to one smaller or equal the input number of channels. For each chunk it is reset to input number of channels and the end it should be equal to the output number of channels.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7879 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoAdding DTX to WebRTC Opus wrapper (relanding).
minyue@webrtc.org [Thu, 11 Dec 2014 16:09:35 +0000 (16:09 +0000)]
Adding DTX to WebRTC Opus wrapper (relanding).

This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.

See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/

Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMerge AEC changes.
pbos@webrtc.org [Thu, 11 Dec 2014 13:46:59 +0000 (13:46 +0000)]
Merge AEC changes.

R=bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/34459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7877 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoWire up RTT statistics to webrtc::Call.
pbos@webrtc.org [Thu, 11 Dec 2014 13:26:09 +0000 (13:26 +0000)]
Wire up RTT statistics to webrtc::Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemove old_factory from WebRtcVideoEngine.
pbos@webrtc.org [Thu, 11 Dec 2014 13:14:30 +0000 (13:14 +0000)]
Remove old_factory from WebRtcVideoEngine.

Minor pending cleanup.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRevert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""
perkj@webrtc.org [Thu, 11 Dec 2014 12:25:57 +0000 (12:25 +0000)]
Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""

Original cl description:

Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMove isolate path into webrtc/build/android/test_runner.py
kjellander@webrtc.org [Thu, 11 Dec 2014 11:59:46 +0000 (11:59 +0000)]
Move isolate path into webrtc/build/android/test_runner.py

This will make it easier to execute tests and allows
for more cleanup in the buildbot recipes.
Now tests can be listed using:
webrtc/build/android/test_runner.py gtest --help
and executed like
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests

TESTED=
Ran:
webrtc/build/android/test_runner.py gtest --help
and verified the tests were listed.
I wiped /sdcard/resources on my device, executed:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests
and verified it passed and that resources/audio_coding/testfile32kHz.pcm
was copied to the device.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7873 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMake an AudioEncoder subclass for PCM16B
henrik.lundin@webrtc.org [Thu, 11 Dec 2014 10:47:19 +0000 (10:47 +0000)]
Make an AudioEncoder subclass for PCM16B

The implementation depends on AudioEncoderPcm to reduce code
duplication.

BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMake an AudioEncoder subclass for iSAC
kwiberg@webrtc.org [Thu, 11 Dec 2014 10:08:19 +0000 (10:08 +0000)]
Make an AudioEncoder subclass for iSAC

BUG=3926

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d