pbos@webrtc.org
896888b7e4
Remove min bitrate from simulcast streams.
...
Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/38529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 15:40:56 +00:00
bjornv@webrtc.org
bac0012120
Extend delay estimation window in AEC to 500 ms on all platforms
...
On non-Android the delay estimator in audio_processing/aec has solely been used for logging purposes. The maximum possible observed delay has been 236 ms. We have seen longer delays for which the delay estimate at best ends up at 236 ms, but can also be 'random'. reported delays are clamped to 500 ms.
This cl extends the delay estimation window to match that.
BUG=4086, 3504, 4113
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7989 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:23:49 +00:00
pbos@webrtc.org
9eacb8cc59
Make P2PTestConductor use VirtualSocketServer.
...
Permits running JsepPeerConnectionP2PTestClient in parallel.
TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*
Review URL: https://webrtc-codereview.appspot.com/37459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:03:19 +00:00
pbos@webrtc.org
c62749fb47
Parallelize MediaRecorder unittests.
...
Exchanging static filenames for temporary ones, permitting tests to be
run in parallel without conflicting parallel uses of the same filenames.
TBR=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 -r 100 out/Debug/libjingle_p2p_unittest
Review URL: https://webrtc-codereview.appspot.com/34589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7987 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:01:20 +00:00
bjornv@webrtc.org
3a70625caf
audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877
...
BUG=N/A
TESTED=Now it builds with aec_debug_dump=1 on Mac
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7986 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-01 22:04:12 +00:00
jiayl@webrtc.org
27f5317560
Use the prod GAE server in AppRTCDemo for iOS.
...
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-31 00:26:20 +00:00
jiayl@webrtc.org
5eb71eb4f4
Fix style issues from lint.
...
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7984 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 22:44:11 +00:00
andrew@webrtc.org
34ac956706
Do not use openmax_dl for MIPS64 platform.
...
This fix is intended for MIPS64 Chromium Android builds, which has no openmax_dl
support at this moment.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31339004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7983 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 18:19:56 +00:00
glaznev@webrtc.org
b2bda67497
Removing old channel code from a few more places.
...
Plus adding peer connection close event.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7982 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 18:15:43 +00:00
pthatcher@webrtc.org
a9b1ec0247
Support for DTLS in OpenSSLAdapter
...
1) Added SetMode() to SSLAdapter and OpenSSLAdapter so the mode can be set to
SSL_MODE_DTLS
2) OpenSSLAdapter overrides SendTo() and RecvFrom() to handle calls from
TurnPort via AsyncUdpSocket
3) OpenSSLAdapter derives from MessageHandler to implement an internal DTLS
timer
4) Updated SSLAdapter unit tests
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7981 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 23:00:14 +00:00
jiayl@webrtc.org
c5fd66dcdf
Accept incoming pings before remote answer is set to reduce connection latency.
...
BUG=4068
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 19:23:37 +00:00
andrew@webrtc.org
84d84471f5
Minor fixes regarding accumulator usage on MIPS platforms.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33729004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7979 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 17:08:44 +00:00
henrika@webrtc.org
b024da3122
Add support for audio device selection in AppRTCDemo.
...
Summary:
- Creates a list of available (possible to select) audio devices.
- Automatically selects (routes audio) the "best/default" audio device.
- If possible, starts a proximity sensor that will switch between headset earpiece and speaker phone based on how close the a person's ear the mobile device is held.
TBR=glaznev
BUG=4103,4109
Review URL: https://webrtc-codereview.appspot.com/31239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7978 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 10:35:06 +00:00
pthatcher@webrtc.org
5ad4178137
Move the Jingle-specific network code into webrtc/libjingle.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-23 22:14:15 +00:00
sprang@webrtc.org
46d4d29a75
Add field trial for screenshare bitrates when using temporal layers.
...
BUG=
R=pbos@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-23 15:19:35 +00:00
mflodman@webrtc.org
1be0a78f45
Removing giles@mozilla.com from WebRTC watchlist.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-22 12:49:14 +00:00
pbos@webrtc.org
53cb74107f
Make RelayServerTest use VirtualSocketServer.
...
Permits running the tests in parallel.
R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w64 out/Debug/rtc_unittests --gtest_filter=RelayServerTest.*
Review URL: https://webrtc-codereview.appspot.com/38479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-22 07:56:42 +00:00
braveyao@webrtc.org
086c8d5a02
Use a temporary buffer to scale a screencast in OnFrameCaptured
...
BUG=3903
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/23909005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-22 05:46:42 +00:00
pthatcher@webrtc.org
4c0544ab07
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
2014-12-19 22:29:55 +00:00
tkchin@webrtc.org
ed1a48b0cd
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
2014-12-19 20:51:02 +00:00
tkchin@webrtc.org
7ce4a584aa
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
2014-12-19 20:47:35 +00:00
aluebs@webrtc.org
ae643ce280
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
2014-12-19 19:57:34 +00:00
stefan@webrtc.org
8817256373
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
2014-12-19 18:00:21 +00:00
stefan@webrtc.org
50f7db8a77
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
2014-12-19 17:55:20 +00:00
jiayl@webrtc.org
a6f7ba6848
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
2014-12-19 17:32:14 +00:00
pbos@webrtc.org
5570769210
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
2014-12-19 15:45:03 +00:00
stefan@webrtc.org
742386a136
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
2014-12-19 15:33:17 +00:00
kwiberg@webrtc.org
aa21f2765b
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
2014-12-19 14:35:57 +00:00
pbos@webrtc.org
d16e839c6d
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
2014-12-19 13:49:55 +00:00
kjellander@webrtc.org
556caffb36
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
2014-12-19 13:28:37 +00:00
stefan@webrtc.org
11d8176cb3
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
2014-12-19 09:52:24 +00:00
pthatcher@webrtc.org
5647877b2d
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
2014-12-19 03:32:59 +00:00
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."
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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.
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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