The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).
For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.
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Review URL: http://webrtc-codereview.appspot.com/239002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed the Windows build.
Fixed whitespace.
Split the platform-specific code for creating a window manager into separate source files since the mac one must be suffixed .mm and not .cc when we happen to use objective-c code. Tested on Linux.
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Review URL: http://webrtc-codereview.appspot.com/214009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@771 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix MediaStreamHandler to make sure it releases the reference to a renderer when it is no longer needed.
Removes the use of the signaling thread in MediaStreamHandler.
Fix renderering in peerconnection_client_dev. It now uses the reference counted render wrapper.
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Review URL: http://webrtc-codereview.appspot.com/242001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@764 4adac7df-926f-26a2-2b94-8c16560cd09d
class VoEAudioProcessing
-API renaming:
SetEchoMetricsStatus() to SetEcMetricsStatus()
GetEchoMetricsStatus() to GetEcMetricsStatus()
since delay logging is not strictly an echo metric.
-New API:
GetEcDelayMetrics()
-Implementations
--SetEcMetricsStatus() sets same status to all EC related metrics, currently Echo Metrics and Delay Logging.
--GetEcMetricsStatus() gets an error if all EC related metrics don't have the same status.
--GetEcDelayMetrics() gets the median and standard deviation of AEC internal delay (on a block by block basis).
class VoECallReport
The changes above leads to changes in the Call Report.
-New API:
GetEcDelaySummary()
-API updates:
ResetCallReportStatistics()
WriteReportToFile()
auto_tests updates:
-Standard test, with new Call Report calls and APM calls
-Extended test, with new Call Report calls and APM calls
Review URL: http://webrtc-codereview.appspot.com/187004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@754 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed merge error.
Fixed cpplint.py warnings.
Fixed presubmit warning.
Whitespace fixes after review.
Rebase from svn.
Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).
The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.
Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.
Fixed a bug which caused test error messages to not get shown.
Added comments to the new test.
Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.
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Review URL: http://webrtc-codereview.appspot.com/188002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@747 4adac7df-926f-26a2-2b94-8c16560cd09d
The code has no intent to be superportable in all possible scenarios, since it will only be used by our own test code.
I reviewed more sophisticated libraries for doing similar things but came to the conclusion that they introduced more dependencies than motivated for this single purpose.
The unit test has been tested successfully executed on Linux (cmd line and Eclipse), Mac (XCode) and Windows (VS2008).
Review URL: http://webrtc-codereview.appspot.com/223002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@734 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed some space issues in vie_autotest_custom_call.cc
Fixed incorrect default codec W&H for I420 in vie_autotest_custom_call.cc
Added functionality to modify a running custom call. The following options were added:
0. Finished modifying custom call
1. Change Video Codec
2. Change Video Size by Common Resolutions
3. Change Video Size by Width & Height
4. Change Video Device
5. Record Incoming Call
6. Record Outgoing Call
7. Play File on Video Channel(Assumes you recorded incoming & outgoing call)
8. Print Call information
Tested with r670, builds fine on Ubuntu & Win7. Mac is not building due to changes in r666, but this patch should be fine on top of it mac as well (compiles fine with r661).
Review URL: http://webrtc-codereview.appspot.com/188003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@728 4adac7df-926f-26a2-2b94-8c16560cd09d
This allows the Mac Make build to pass. We were hacking it in XCode with "-x objective-c++", but gyp/Make doesn't seem to accept that flag.
Also switch Objective-C #includes to #imports.
There is one file missing from this: vie_autotest_main.cc, because it's required on multiple platforms. I'm not immediately sure what the best approach is there, but the Objective-C headers should be somehow hidden.
Review URL: http://webrtc-codereview.appspot.com/153005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@726 4adac7df-926f-26a2-2b94-8c16560cd09d
