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Author SHA1 Message Date
pbos@webrtc.org
7ee822805d Remove TEXT(x) for BUILDINFO macros. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1453004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 09:29:03 +00:00
fischman@webrtc.org
d6ed000585 This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build. 
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1444005

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3999 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-10 16:34:01 +00:00
fischman@webrtc.org
6a36f0e46f Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation. 
BUG=webrtc:1741

TEST=Build and run the Android WebRTC demo application
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1439006

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 17:40:33 +00:00
braveyao@webrtc.org
e525309004 WebRTCDemo Android doesn't hangle activity recreation correctly. 
Also optimize Statsview a little bit.

BUG=1740
TEST=Manual test with WebRTCDemo Android
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 08:52:50 +00:00
braveyao@webrtc.org
ebdfa8dcba Add fischman into OWNERS of WebRTCDemo Android. 
BUG=
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 07:30:38 +00:00
andrew@webrtc.org
d72262dc01 Fix compile errors in ViE with latest clang. 
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:

error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
 VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
  VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~
                              static_cast

This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).

Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
        AutoTestSleep(std::numeric_limits<long>::max());
        ~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

This fixes the errors and is required before stable can be rolled in Chromium.

TBR=mflodman,andresp

Review URL: https://webrtc-codereview.appspot.com/1450004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 02:12:07 +00:00
andresp@webrtc.org
44272739c2 Clean creation of VideoEngine: 
- clean a static variable just used to debug and not so necessary IMO.
 - clean a really ugly reinterpret cast
 - clean a extern "C" code and loading of dlls which is no longer in use.

Review URL: https://webrtc-codereview.appspot.com/1385006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 19:20:23 +00:00
stefan@webrtc.org
ef14488d03 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. 
BUG=1663
R=mikhal@webrtc.org, ronghuawu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 19:16:33 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
elham@webrtc.org
df3da84ec8 Updated WebRTC version number to 3.30 
R=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1404005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 23:11:37 +00:00
solenberg@webrtc.org
2580bc4c30 Get rid of some unnecessary copying when sending REMBs. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1325005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 09:22:14 +00:00
pwestin@webrtc.org
42636e82d0 Removing bad code resulting in flaky test. 
BUG=1723
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 21:02:04 +00:00
pwestin@webrtc.org
52b4e8871a Adding trace and changing pacing constants 
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 19:02:17 +00:00
pwestin@webrtc.org
0d95e06a2f Bugfix custom call stop. 
BUG=1717
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1388004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 18:25:03 +00:00
braveyao@webrtc.org
3c48f31e5b WebRTCDemo Android app to route audio to headphone when it's plugged in. 
BUG=1654
TEST=WebRTCDemo app

Review URL: https://webrtc-codereview.appspot.com/1348004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 03:18:00 +00:00
andrew@webrtc.org
342353780d Consolidate common_audio into a single target. 
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.

R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:43:26 +00:00
pwestin@webrtc.org
d35964a1ce Fixing AV sync. 
Increased 2 const to allow for a bigger difference in AV sync.

BUG=1711

Re-wrote the ComputeDelays to be readable and remove the possibilities of returning values lower than base_target_delay_ms

R=mflodman@webrtc.org, mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 16:06:10 +00:00
mikhal@webrtc.org
dd807ac474 Adding buffered mode to loopback test 
Review URL: https://webrtc-codereview.appspot.com/1371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:19:47 +00:00
mikhal@webrtc.org
47128ab5ab Removing vie file related code from vie_custom_call 
Follow up on https://code.google.com/p/webrtc/source/detail?r=3900

Review URL: https://webrtc-codereview.appspot.com/1361004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 20:09:54 +00:00
pwestin@webrtc.org
4e545b33b3 Fixed remaining nits from Stefan 
Review URL: https://webrtc-codereview.appspot.com/1323007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 15:23:34 +00:00
pwestin@webrtc.org
91563e42da Fix the encoder pause logic. 
BUG=1691

Review URL: https://webrtc-codereview.appspot.com/1352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 22:20:08 +00:00
mikhal@webrtc.org
b84f13f185 Disabling avi file interface 
Review URL: https://webrtc-codereview.appspot.com/1351004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 18:07:32 +00:00
stefan@webrtc.org
8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." 
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.

BUG=1613
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1327008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:48:32 +00:00
stefan@webrtc.org
ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. 
BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 15:58:23 +00:00
pwestin@webrtc.org
63117339dc Updated the sync module with a slow moving filter 
Review URL: https://webrtc-codereview.appspot.com/1326008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 18:57:14 +00:00
mflodman@webrtc.org
7c9e992d05 Removed unused variable. 
Review URL: https://webrtc-codereview.appspot.com/1320013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 13:05:41 +00:00
mflodman@webrtc.org
aeff4f3003 Fixing Coverity issues. 
BUG=C14457, C10611

Review URL: https://webrtc-codereview.appspot.com/1320012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 12:41:57 +00:00
kjellander@webrtc.org
c41478f7eb Ensure build_demo.py run subprocesses with bash shell. 
It turns out the default shell becomes /bin/sh on Lucid. By specifying the shell for subprocess.check_call we ensure bash is used.

Thanks to yujie.mao@intel.com for pointing this out.

BUG=1659
TEST=Successful build with build_demo.py both on Ubuntu Lucid and Precise.
TBR=leozwang

Review URL: https://webrtc-codereview.appspot.com/1343004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 11:50:42 +00:00
mflodman@webrtc.org
65f995a3df New ViE interface. 
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/1113004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 12:02:52 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files 
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 12:40:34 +00:00
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
stefan@webrtc.org
b8e7f4cc97 Change capture interface to use NTP capture time. 
Move NTP functionality to Clock.

BUG=1563
TEST=trybots and vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/1313005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 11:56:23 +00:00
pwestin@webrtc.org
1de01354e6 Adding playout buffer status to the voe video sync 
Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 20:23:35 +00:00
elham@webrtc.org
1b2a6e0be0 Updated WebRTC version number to 3.29 
TBR=mallinath1 
Review URL: https://webrtc-codereview.appspot.com/1305005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 21:28:33 +00:00
fischman@webrtc.org
6f41ca9fd2 WebRTCDemo: Enable making multiple calls. 
Previously after the first call subsequent attempts to bind the RTP/RTCP ports would fail, since r3754.

BUG=1618

Review URL: https://webrtc-codereview.appspot.com/1302007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 20:33:27 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
29758de9b6 Always set render delay in ViEChannel::RegisterExternalDecoder. 
BUG=1523

Review URL: https://webrtc-codereview.appspot.com/1219007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:34:42 +00:00
pwestin@webrtc.org
6faf71d27b Remove the old unused udp_transport 
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 23:25:25 +00:00
mflodman@webrtc.org
367804cce2 Clean packets on the network when closing + made loopback test actually run again. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1290006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:42:50 +00:00
kjellander@webrtc.org
10eb92039b Add GYP target for WebRTC Video demo for Android. 
Add a build target for the Video demo app for Android that only
exists when OS=='android' during build.

Note that this doesn't solve webrtc:1029, it's more like a workaround
waiting for the complete solution, which is to great a proper GYP target
that doesn't involve an action and an external script.

BUG=1029
TEST=Built successfully with:
source build/android/envsetup.sh
gclient runhooks
ninja -C out/Debug
Also verified the target is not present when OS is not 'android'.

Review URL: https://webrtc-codereview.appspot.com/1286004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:36:32 +00:00
pbos@webrtc.org
b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. 
BUG=1575

Review URL: https://webrtc-codereview.appspot.com/1282005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:27:38 +00:00
pwestin@webrtc.org
b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use 
Review URL: https://webrtc-codereview.appspot.com/1203009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:51:42 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
pwestin@webrtc.org
82dcc9ff11 Remove UDP transport API from ViE 
Review URL: https://webrtc-codereview.appspot.com/1232004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 20:37:14 +00:00
elham@webrtc.org
65243bdb18 Updated Webrtc version to 3.28 
Review URL: https://webrtc-codereview.appspot.com/1272006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 16:17:26 +00:00
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +00:00
leozwang@webrtc.org
458194ba65 Fix broken audio. 
The problem was introduced in 3712, no need to external transport in
real test app, revert the change.

TBR=pwestin@webrtc.org
BUG=1539
Review URL: https://webrtc-codereview.appspot.com/1266005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:55:54 +00:00