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Author SHA1 Message Date
mflodman@webrtc.org
21beaf97e7 Adding Stefan as VideoEngine owner, removing Per. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1762004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-04 12:29:08 +00:00
braveyao@webrtc.org
0b8636a783 In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. 
BUG=
TEST=manual Test
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-04 07:24:12 +00:00
henrike@webrtc.org
1303af31d6 Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. 
Alternative solution to http://webrtc-codereview.appspot.com/1748004/.

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 21:50:33 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
tina.legrand@webrtc.org
45426eadf5 In call to Opus decoder: frame length too large 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1201
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1752004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 13:32:04 +00:00
tina.legrand@webrtc.org
f6f033f8bd Possible divide by 0 in ACM. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1551
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1757004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 12:00:14 +00:00
tina.legrand@webrtc.org
b1698ab827 Error in update of read index in ACM 
Fixing a bug where we increase read index with too few samples when the input is stereo.

BUG=https://code.google.com/p/webrtc/issues/detail?id=714
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1753004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 09:25:34 +00:00
tommi@webrtc.org
ecd3c800c4 Add Magnus to root owners. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1752005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 08:21:41 +00:00
pbos@webrtc.org
c66aaaf921 Rename unit_test.{cc,h} under module_unittest. 
Squelches the following Windows trybot warning:
warning LNK4042: object specified more than once; extras ignored

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

Review URL: https://webrtc-codereview.appspot.com/1758004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 07:56:33 +00:00
yujie.mao@webrtc.org
510dfad636 Update myself in webrtc watchlist 
BUG=NONE
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 01:13:18 +00:00
pbos@webrtc.org
65a1f2cb2b Remove log of undefined input values in GetCodec. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1755004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-02 13:02:14 +00:00
pbos@webrtc.org
504af45a6f Diff NTP and internal once in VideoCaptureImpl. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1754004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-02 10:15:43 +00:00
fischman@webrtc.org
546c91dc2e Build all java files into jar for each module on Android 
BUG=None
TEST=All java files in each module are built into jar and used by WebRTCDemo app
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1696004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-01 17:52:39 +00:00
yujie.mao@webrtc.org
d4803ced60 WebRTCViEDemo: Use global reference when passing variables across different threads 
There are JNI local reference changes in ICS when Android SDK
target level API >= 14.
http://android-developers.blogspot.com/2011/11/jni-local-reference-changes-in-ics.html

BUG=NONE
TEST=WebRTCViEDemo works well using MediaCodec Decoder/Renderer
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1744004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-01 14:55:37 +00:00
braveyao@webrtc.org
90cc3b95b7 Android opengles renderer: add thread sync to swap frame and draw native. 
BUG=1616
TEST=Manual Test
R=fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1738005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-28 23:53:11 +00:00
hclam@chromium.org
5616abadf5 Suppress excessive logging in video_coding 
Only prints the warning message if a frame was dropped.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1735004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 19:47:40 +00:00
henrike@webrtc.org
2a7fd5355d Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. 
BUG=N/A
R=andrew@webrtc.org, kjellander@google.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1730004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 18:36:28 +00:00
henrike@webrtc.org
83cebb25d7 Removes unused main function that is poluting the build. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1728005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 18:31:13 +00:00
fischman@webrtc.org
0021632f40 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! 
BUG=1980
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1734004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 17:35:32 +00:00
fischman@webrtc.org
1d4a2d5daf Move TickTime::QueryOsForTicks out-of-line 
This inline function is no longer expanded on arm Android, but on x86 Android it
will still be expanded. Move it out-of-line to make things consistent.

This change list will also fix a potential bug on webrtc for Android:
Since the inline function won't be expanded on arm Android,
TickTime::MillisecondTimestamp and Clock::GetRealTimeClock()->TimeInMilliseconds
will be treated as function call, due to macro WEBRTC_CLOCK_TYPE_REALTIME's
guard defined in system_wrappers module they will get current time using
CLOCK_REALTIME.

But on x86 Android, the inline function will be expanded to where it's been
called, if the call happens in other compilation units which don't have
WEBRTC_CLOCK_TYPE_REALTIME definition, it will get current time using
CLOCK_MONOTONIC, while Clock::GetRealTimeClock()->TimeInMilliseconds will always
use CLOCK_REALTIME, then there will be two types of time in x86 Android which
will cause some weird issues like all received remote streams will be dropped
due to future render timestamp.

BUG=None
TEST=WebRTCViEDemo application works well on both arm and x86 Android
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1688004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 17:15:20 +00:00
stefan@webrtc.org
4cf1a8af69 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. 
The idea is to have all frames not in use be stored in free_frames_, and whenever a packet from a new frame arrives we can just pop a frame from free_frames_. When a frame is grabbed for decoding it will be removed from all lists, and will be added to free_frames_ when it's returned to the jitter buffer.

We should be able to remove the state enum completely later, as their state is defined by the list they are in. But I'll keep it around for now to simplify the cl.

TEST=try bots and vie_auto_test --automated
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1721004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 15:20:14 +00:00
phoglund@webrtc.org
7bcc7e3b43 Fixed bad parameter passing in compare_videos.py 
BUG=http://crbug.com/254932
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1733004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 14:05:26 +00:00
pbos@webrtc.org
2de80ddc72 Fix unnamed-type-template-args warnings on clang. 
BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1732004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 10:18:09 +00:00
fischman@webrtc.org
3145a642b7 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. 
BUG=1980
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 20:20:05 +00:00
mflodman@webrtc.org
e6168f5f41 Adding a first simple version of overuse detection, but not hooked up. 
BUG=
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 11:23:01 +00:00
mflodman@webrtc.org
1c986e7c89 Removed ViE file API. 
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 09:12:49 +00:00
solenberg@webrtc.org
a5fd2f1348 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1697004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:36:07 +00:00
solenberg@webrtc.org
892d750ba6 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1698004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:22:53 +00:00
solenberg@webrtc.org
91811e2b04 Remove unused multi stream bandwidth estimator. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1712004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 20:36:14 +00:00
stefan@webrtc.org
a4c5abb52a Make sure padding packets are sent. 
BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 15:46:16 +00:00
vikasmarwaha@webrtc.org
bb25256775 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. 
R=dutton@google.com, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/1627006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 14:52:51 +00:00
sergeyu@chromium.org
3348ae2b97 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. 
kCGLPFAFullScreen is marked deprecated starting with 10.6 in the 10.9 SDK,
but it's functional on 10.6 and this code only runs on 10.6 and will go away
when support for 10.6 is dropped.

BUG=webrtc:1958
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1710004

Patch from Nico Weber <thakis@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-21 23:33:10 +00:00
marpan@webrtc.org
bb4f225a5b Roll libvpx to 207593. 
-pick up libvpx roll to c259af4f.

TBR: ajm@google.com

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1707004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-21 22:19:34 +00:00
hclam@chromium.org
6eb53f71d6 Fix memory bot failure 
Exit the method with critical setting held. This should make
the memory bot happy.

TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1704005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 23:01:39 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
mikhal@webrtc.org
9ca7360b97 VCM: removing max jitter estimate 
BUG= 1921
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1690004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:13:07 +00:00
andrew@webrtc.org
0851df8d60 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. 
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.

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

Review URL: https://webrtc-codereview.appspot.com/1683004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 17:03:47 +00:00
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing. 
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 14:13:42 +00:00
kjellander@webrtc.org
2d7617afce Add dummy Android test APK to be used for buildbot automation testing. 
Until we have WebRTC test targets created for Android, this test
makes it possible to move forward for buildbot automation.

TEST=Android NDK buildbot and local execution of:
source build/android/envsetup.sh
gclient runhooks
ninjar -C out/Debug
verified the out/Debug/simple_apk dir exists and has the files.
BUG=1882
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1688005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 09:10:49 +00:00
fbarchard@google.com
d7148c86c5 Use 3 threads for higher than 720p resolutions 
BUG=1893
TEST=untested
R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1684004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 22:06:42 +00:00
hclam@chromium.org
30fb7b83d5 Add a log message to see video delay break down 
Shows video delay in terms of:
1. Min playout delay
2. Jitter delay
3. Max decode time
4. Render delay

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1674004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 21:37:09 +00:00
kjellander@webrtc.org
6cfe178af2 Chromium Android tools for test execution. 
The md5sum and forwarder2 binaries from Chromium's
src/tools/android are needed to be able to run tests using the
test framework launched by build/android/run_tests.py.
Since they depend on Chromium's base, we're using a precompiled
copy for WebRTC's purposes.

Linux works out of the box if Chromium's Android build instructions
at https://code.google.com/p/chromium/wiki/AndroidBuildInstructions
are used. Mac runs into problems earlier in the build toolchain,
but as Mac is not a supported Android development platform in Chrome,
the files will have to be copied manually on that platform for now.

TEST=Synced, built and ran a test APK using run_tests.py.
BUG=1882
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 07:14:33 +00:00
sergeyu@chromium.org
a20eb91154 Make ScreenCapturerMac work in versions of OSX before Lion. 
The screen capturer was broken when moving code to webrtc: width
and height parameters for glReadPixels were swapped by mistkake.

BUG=crbug.com/244102
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1678005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 22:22:40 +00:00
sergeyu@chromium.org
9e182795a9 Enable ScreenCapturer unittests 
previously ScreenCapturer unittests were disabled by mistake

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 21:14:36 +00:00
sergeyu@chromium.org
a590b41c9a Use intptr_t to represent window IDs on all platforms. 
Previously void* was used on windows which makes it harder to work
with the IDs in cross-platform code.

R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1672004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 20:02:21 +00:00
stefan@webrtc.org
508a84b255 Wire up pacer-based padding. 
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.

Padding will for now only be generated by the first sending RTP module.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 12:53:37 +00:00
stefan@webrtc.org
50fb4afade Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1678004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 07:33:58 +00:00
stefan@webrtc.org
c8b29a2feb Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1677004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 07:13:16 +00:00
hclam@chromium.org
7262ad1385 Fix AV sync issue 
r4229 introduced an AV sync issue due to an error.
This is a one linear fix and provides the correct
current video delay for synchronization.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1675004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-15 06:51:27 +00:00
hclam@chromium.org
9b23ecb939 Log current and target AV delay in ViESyncModule 
R=mikhal@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1668006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 23:30:58 +00:00