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Author SHA1 Message Date
pbos@webrtc.org
f2e7bc6b6a Added maxlen=80 to CheckLongLines() call in PRESUBMIT.py 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1285006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 15:46:07 +00:00
pbos@webrtc.org
034f004a4f WebRtc_Word32 => int32_t in video_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1203008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:13:29 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +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
pbos@webrtc.org
ff7e1303e8 WebRtc_Word32 => int32_t remote_bitrate_estimator/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1275009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:04:37 +00:00
hta@webrtc.org
37bf5847dc Show stats from both sides 
This change shows the stats generated both at the sending PeerConnection
and at the receiving PeerConnection.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1290005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 10:05:55 +00:00
vikasmarwaha@webrtc.org
222e9948f5 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. 
Review URL: https://webrtc-codereview.appspot.com/1291004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 05:58:15 +00:00
wu@webrtc.org
123b618f48 Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. 
BUG=crbug.com/226301
TBR=henrike
Review URL: https://webrtc-codereview.appspot.com/1293004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 04:13:51 +00:00
turaj@webrtc.org
2e6b7e938f In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. 
test=try bots.
Review URL: https://webrtc-codereview.appspot.com/1272004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 00:08:11 +00:00
henrika@webrtc.org
19da719a5f Resolves TSan v2 reports data races in voe_auto_test. 
--- Note that I will add more fixes to this CL ---

BUG=1590

Review URL: https://webrtc-codereview.appspot.com/1286005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 14:34:57 +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
835dbf4516 Fix no received audio in tests. 
BUG=1582, 1581
Review URL: https://webrtc-codereview.appspot.com/1281005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 17:24:15 +00:00
henrika@webrtc.org
aa527bbc91 Disabling MixingTests due to race conditions. 
BUG=1580
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1285005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 15:19:10 +00:00
hta@webrtc.org
fcb7c38b15 Two more sleep calls converted to use SleepMs(). 
This is CL 753005 in its new home.

BUG=603

Review URL: https://webrtc-codereview.appspot.com/1201008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:56:34 +00:00
henrika@webrtc.org
bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC 
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:39:09 +00:00
pwestin@webrtc.org
0c45957e3a Remove UDP transport API from VoE 
Review URL: https://webrtc-codereview.appspot.com/1236004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 15:43:57 +00:00
henrika@webrtc.org
0746ce1465 Fixes memory leak in AudioLevel class reported by memory try bots. 
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1275008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 11:58:12 +00:00
henrika@webrtc.org
d108a46206 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer 
BUG=225690

Review URL: https://webrtc-codereview.appspot.com/1269008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 11:25:31 +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
pbos@webrtc.org
7b859cc1e9 Webrtc_Word32 => int32_t in video_coding/main/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 15:54:38 +00:00
henrike@webrtc.org
cfc07c943f Revert of r3747. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:55:44 +00:00
hta@webrtc.org
95d88735ee Two more sleep calls converted to use SleepMs(). 
BUG=603

Review URL: https://webrtc-codereview.appspot.com/753005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:46:33 +00:00
henrika@webrtc.org
4ff956f428 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer 
BUG=159112

Review URL: https://webrtc-codereview.appspot.com/1201007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 11:59:11 +00:00
kjellander@webrtc.org
46e626d3b8 Fix gflags compile error on x86 Android 
This CL is the landing of http://review.webrtc.org/1277004/ for yujie.mao@intel.com.

I verified the added files are identical with the previously added ones
in third_party/google-gflags/gen/arch/linux/ia32 (which is the way this library needs to be handled when supporting the additional Android platforms).

BUG=none
TEST=Successfully compiled WebRTC on Linux Precise with:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug

Review URL: https://webrtc-codereview.appspot.com/1273005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 11:07:04 +00:00
justinlin@chromium.org
f81fad6267 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher 
than 2^16kbps.
Review URL: https://webrtc-codereview.appspot.com/1275004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 22:25:11 +00:00
fbarchard@google.com
747c4cc96e For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. 
BUG=none
TEST=run a hangout and screencast high framerate, high resolution windows of youtube.  Observe that 1 cpu is insufficient to maintain high framerate with complex content.
Review URL: https://webrtc-codereview.appspot.com/1203006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 22:16:45 +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
marpan@webrtc.org
7f6b7cbcfc Revert r3743. 
TBR=andrew@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1272005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-29 21:35:22 +00:00
marpan@webrtc.org
e882a47c8d Roll libvpx to 191157. 
-Pick up the libvpx roll to 8015a9ae.

TBR=andrew@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1273004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-29 21:16:24 +00:00
marpan@webrtc.org
29f34b8727 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 
Review URL: https://webrtc-codereview.appspot.com/1270004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 18:57:46 +00:00
henrike@webrtc.org
626c663115 Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 16:31:51 +00:00
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +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
wu@webrtc.org
80fccc29de Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." 
> Removed CPU APIs from VoEHardware. Code is now only used by test applications.
> 
> BUG=8404677
> 
> Review URL: https://webrtc-codereview.appspot.com/1238004

TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 23:38:21 +00:00
henrike@webrtc.org
4c138e8fca Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
BUG=8404677

Review URL: https://webrtc-codereview.appspot.com/1238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 21:23:42 +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
turaj@webrtc.org
4b1cd5c5c0 G722-stereo has been missing when creating AudioDecoder. 
Review URL: https://webrtc-codereview.appspot.com/1266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:42:48 +00:00
turaj@webrtc.org
4d06db557a NetEq4 fails if the first packets inserted in are out-of-band DTMFs. 
I had to take few steps to solve this issue. I have comments on places I made cahanges to clarify why I did the change.

   
Review URL: https://webrtc-codereview.appspot.com/1195004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 18:31:42 +00:00
stefan@webrtc.org
e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. 
TBR=pwestin@webrtc.org

BUG=1524

Review URL: https://webrtc-codereview.appspot.com/1261005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 17:01:48 +00:00
fischman@webrtc.org
add50b94a5 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. 
(required bumping minSdkVersion to 14)

This fixes a RuntimeException thrown on GalaxyNexus (but not N7, N4, or NS)
during startPreview() after the sequence of Start(), Stop(), Start(); seemingly
GN's OMX stack can't deal with parallel startPreview() & setPreviewDisplay() in
this situation.

Also:
- Only set the surface in the camera when valid
- Remove duplicate assignment
- Fix error check on voiceChannel allocation to account for multiple channel creation due to orientation change causing onDestroy()/onCreate() on the app, and rampant use of process-static holders for VoE data.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:48:34 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
henrike@webrtc.org
686001dd96 Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events). 
Note that this means that there is no new code. The code has been taken directly from condition_variable_win.cc/h compensating minimally to be able to split up the two code paths.

Tested by:
1) Disabling native implementation and send to try bots.
2) Only return native implementation (i.e. if native implementation returns NULL there will be a crash when using the condition variable) and send to try bots.
3) The final cl sent to trybots.
All tests pass.

The changes are due to static analyzer code complaints.

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:16:05 +00:00
andrew@webrtc.org
1b31c78e5f Remove VoE's default call in Trace::SetLevelFilter. 
This is an application level setting. Applying it here has the potential to override the application's preferences.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:09:48 +00:00
solenberg@webrtc.org
d8a6e72057 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1232005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:02:30 +00:00
andrew@webrtc.org
0633cccb4f Alphabetize include order in fake_voe_external_media.h. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 01:57:24 +00:00
fischman@webrtc.org
0e3077ab1f Restart Android capture after orientation change. 
Also prevent an NPE on exit.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 22:08:51 +00:00
andrew@webrtc.org
c83a00ad49 Add some VoE and AudioProcessing mocks. 
Includes a bit of shared helpers in fake_common.h.

Review URL: https://webrtc-codereview.appspot.com/1221004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 21:20:38 +00:00