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Author SHA1 Message Date
aluebs@webrtc.org
0c39e91cc8 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
 2014-12-18 22:22:04 +00:00
andrew@webrtc.org
1090a6eccf 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
 2014-12-18 21:36:18 +00:00
pthatcher@webrtc.org
aacc23465b 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
 2014-12-18 20:31:29 +00:00
jiayl@webrtc.org
16a05dddb8 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
 2014-12-18 20:12:03 +00:00
pthatcher@webrtc.org
f5847d7746 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
 2014-12-18 17:09:11 +00:00
asapersson@webrtc.org
cb79141eab 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
 2014-12-18 14:30:32 +00:00
pbos@webrtc.org
ce4e9a3562 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
 2014-12-18 13:50:16 +00:00
pbos@webrtc.org
98c04b38a8 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
 2014-12-18 13:12:52 +00:00
sprang@webrtc.org
9b79197c80 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
 2014-12-18 11:53:59 +00:00
pbos@webrtc.org
f832a6d090 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
 2014-12-18 09:56:09 +00:00
henrik.lundin@webrtc.org
eed7a22bbf 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
 2014-12-18 09:52:36 +00:00
pbos@webrtc.org
dd8f6f3d48 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
 2014-12-18 09:18:42 +00:00
pbos@webrtc.org
a9cf079248 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
 2014-12-18 09:12:21 +00:00
pbos@webrtc.org
e468bc9e60 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
 2014-12-18 09:11:33 +00:00
henrik.lundin@webrtc.org
cab1291745 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
 2014-12-18 06:58:42 +00:00
guoweis@webrtc.org
4fba293c87 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
 2014-12-18 04:45:05 +00:00
pthatcher@webrtc.org
4cb3856a4d 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
 2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58 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
 2014-12-18 01:22:02 +00:00
guoweis@webrtc.org
c51fb9348d 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
 2014-12-18 00:30:55 +00:00
andrew@webrtc.org
0ab42bc3f6 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
 2014-12-17 22:56:09 +00:00
pthatcher@webrtc.org
bc03192560 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
 2014-12-17 22:15:11 +00:00
guoweis@webrtc.org
0eb6eec5cb 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
 2014-12-17 22:03:33 +00:00
aluebs@webrtc.org
6f10ae25ea 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
 2014-12-17 17:28:31 +00:00
pbos@webrtc.org
eb544460e4 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
 2014-12-17 15:23:29 +00:00
tommi@webrtc.org
209df9bf77 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
 2014-12-17 14:09:05 +00:00
pbos@webrtc.org
e728ee03ba 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
 2014-12-17 13:43:55 +00:00
tommi@webrtc.org
5263c58923 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
 2014-12-17 12:35:29 +00:00
sprang@webrtc.org
70f74f3f7b 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
 2014-12-17 10:57:10 +00:00
asapersson@webrtc.org
45a272ab22 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
 2014-12-17 10:27:57 +00:00
kwiberg@webrtc.org
e102e8147b 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
 2014-12-17 07:30:23 +00:00
braveyao@webrtc.org
38881be912 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
 2014-12-17 05:59:41 +00:00
guoweis@webrtc.org
950c518251 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
 2014-12-16 23:01:31 +00:00
andrew@webrtc.org
971bf557e2 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
 2014-12-16 22:28:20 +00:00
pthatcher@webrtc.org
f050791ba0 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
 2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c 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
 2014-12-16 21:37:37 +00:00
pthatcher@webrtc.org
e2b7585bc2 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
 2014-12-16 21:09:08 +00:00
henrik.lundin@webrtc.org
a32487f97b 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
 2014-12-16 21:04:55 +00:00
pthatcher@webrtc.org
02c21dbef1 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
 2014-12-16 21:04:41 +00:00
andrew@webrtc.org
08df9b2841 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
 2014-12-16 20:57:15 +00:00
aluebs@webrtc.org
cf6d0b64ef 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
 2014-12-16 20:56:09 +00:00
andrew@webrtc.org
2510d11c0f 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
 2014-12-16 20:47:42 +00:00
andrew@webrtc.org
048c5029f5 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
 2014-12-16 20:17:21 +00:00
pbos@webrtc.org
451a133f44 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
 2014-12-16 14:48:47 +00:00
henrik.lundin@webrtc.org
c1c9291e9b 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
 2014-12-16 13:41:36 +00:00
kwiberg@webrtc.org
88bdec8c3a 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
 2014-12-16 12:49:37 +00:00
kjellander@webrtc.org
0198933b3d 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
 2014-12-16 12:29:59 +00:00
asapersson@webrtc.org
d08d389ce8 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
 2014-12-16 12:03:11 +00:00
bjornv@webrtc.org
b395a5ea65 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
 2014-12-16 10:38:10 +00:00
guoweis@webrtc.org
55360ae402 Revert "Add adapter_type into Candidate object." 
This reverts commit aaf02cc2d4.

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
 2014-12-16 05:28:10 +00:00
marpan@webrtc.org
d021bbbc9e 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
 2014-12-16 00:21:47 +00:00