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Author SHA1 Message Date
wu@webrtc.org
f3f2f6abdb * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM. 
* Split the WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER into WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE and WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER.
* Add DummyDeviceInfo for the case when WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE is not defined.
Review URL: http://webrtc-codereview.appspot.com/224005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:42:17 +00:00
henrike@webrtc.org
509c9c5d09 operator + is evaluated before ?: 
Parenthesis ensures the intended behavior.
Review URL: http://webrtc-codereview.appspot.com/239003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:31:01 +00:00
henrike@webrtc.org
4df8c9a2ed Review URL: http://webrtc-codereview.appspot.com/243001 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:30:25 +00:00
andrew@webrtc.org
7ecdf585cb Enable chromium_code:1 in the Chrome build. 
Review URL: http://webrtc-codereview.appspot.com/240001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 17:53:56 +00:00
stefan@webrtc.org
ffd28f95c5 Request key frames to battle error propagation. 
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.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:55:39 +00:00
mikhal@webrtc.org
d0752c370d video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. 
Review URL: http://webrtc-codereview.appspot.com/245001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:48:30 +00:00
mflodman@webrtc.org
c693bac6e7 Only start ViEPerformanceMonitor when needed. 
Tested by taking the added part in base extended test and running in Standard test with cpu threashold in ViEPeroformanceMonitor manually changed to 0.

Review URL: http://webrtc-codereview.appspot.com/240005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 13:40:58 +00:00
phoglund@webrtc.org
b5475d0076 vie_auto_test will now obey the Mac .mm rules for files including objective-c code. 
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.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/214009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 10:59:39 +00:00
bjornv@webrtc.org
4c636764b7 Updated the AEC delay logging to output values in ms. PB output updated. 
Review URL: http://webrtc-codereview.appspot.com/223003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 08:47:40 +00:00
mflodman@webrtc.org
cc412c1735 Remove second instance of ViE PerformanceMonitor. 
Review URL: http://webrtc-codereview.appspot.com/244001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 08:27:30 +00:00
mflodman@webrtc.org
ce8813da4e Using id instead of name when setting Mac/QTKit capture device. 
Review URL: http://webrtc-codereview.appspot.com/241002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 06:45:16 +00:00
andrew@webrtc.org
4d5d5c1267 Reorganize the audio_processing source. 
- Remove main and source directories.
- Change .gyp, .gypi and Android.mk files correspondingly. No other
  source changes.

Review URL: http://webrtc-codereview.appspot.com/241001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 01:40:33 +00:00
andrew@webrtc.org
5d3bdf71ab Fix clang warnings in ViE autotest. 
Review URL: http://webrtc-codereview.appspot.com/239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 01:09:41 +00:00
wu@webrtc.org
8fd93d4d96 Move DeliverCapturedFrame from private to protected. 
Review URL: http://webrtc-codereview.appspot.com/246001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 00:16:36 +00:00
perkj@webrtc.org
1305a1d05e Fix rendering in new PeerConnection API. 
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.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 11:54:46 +00:00
bjornv@webrtc.org
52eddf7378 Made Tina, Andrew and Jan as OWNERS to entire common_audio and removed the sub-OWNERS files. Let me know if that's fine. 
Review URL: http://webrtc-codereview.appspot.com/225006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 07:57:04 +00:00
stefan@webrtc.org
5b15cfc6dd Fix BWE unit test build issue 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 07:22:33 +00:00
kjellander@webrtc.org
61f07c3184 I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files. 
The ApmTest.Process test is still failing and needs to be resolved.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/194002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 06:54:58 +00:00
henrik.lundin@webrtc.org
5dedd0ee38 Handling of white-space in DataLog::Combine 
The Combine method cannot handle white-space. Adding a comment to
the header file saying this, and modifying the unittests. Also,
adding a new unittest to test the method.

Review URL: http://webrtc-codereview.appspot.com/217001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 05:45:08 +00:00
amyfong@webrtc.org
929789b528 vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call 
- moved all of the custom call specific functions out of vie_autotest.h and into vie_autotest_custom_call.cc
- added option to modify a running call's video protection method
Review URL: http://webrtc-codereview.appspot.com/234001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@759 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:57:08 +00:00
wu@webrtc.org
76aea651ff When _audioConfigured, should not try to use the _video. 
Review URL: http://webrtc-codereview.appspot.com/224004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@758 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:40:32 +00:00
henrike@webrtc.org
0d55c8f96d Adding peerconnection_unittest. 
Review URL: http://webrtc-codereview.appspot.com/226004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:12:45 +00:00
mallinath@webrtc.org
5cb3064642 The change will separate the media tracks based on media type. MediaStreamInterface currently will have list for audio and video. This way we don't need to check for the track type before converting to respective mediatrack. 
Review URL: http://webrtc-codereview.appspot.com/230003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 13:19:08 +00:00
perkj@webrtc.org
63257d4bd2 Implement proxy for both audio and video tracks. 
The purpose of the proxy is that all calls to MediaStreamTracks should be done on the signaling thread.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/225004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 11:39:09 +00:00
bjornv@webrtc.org
3765bd2cc2 Added AEC delay logging metrics to VoE. Echo metrics and delay logging metrics are enabled simultaneously through the SetEcMetricsStatus(). Updated standard and extended VoE tests. 
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
 2011-10-17 08:49:23 +00:00
wu@webrtc.org
f10ea31211 Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes. 
Review URL: http://webrtc-codereview.appspot.com/219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 17:16:04 +00:00
marpan@webrtc.org
14aaaf116a Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming. 
Review URL: http://webrtc-codereview.appspot.com/231001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 16:28:02 +00:00
wu@webrtc.org
55c39f0940 Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office. 
Review URL: http://webrtc-codereview.appspot.com/230001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 15:34:19 +00:00
wu@webrtc.org
58691ebb97 Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.) 
Review URL: http://webrtc-codereview.appspot.com/229001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 15:13:16 +00:00
stefan@webrtc.org
d0bdab0128 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. 
Also adding tests for this in vie_auto_test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/199001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 14:24:54 +00:00
hta@webrtc.org
e698eb7e27 Make the sanity check test a little more robust, and add a README file. 
Review URL: http://webrtc-codereview.appspot.com/220006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 13:56:26 +00:00
phoglund@webrtc.org
26c041673f Added more tests, fixed a bug and refactored. 
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.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/188002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 13:00:20 +00:00
bjornv@webrtc.org
2111d3b0b0 Removed the vad_const files and added the constants to the files where they are 
used. Having them in a separate file did not add anything in readability or
conceptual overview.
Review URL: http://webrtc-codereview.appspot.com/230004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 12:58:34 +00:00
bjornv@webrtc.org
a59d80db45 Updated fixed point output data file after changes in nsx. Verified bitexactness before that CL and the CLs afterwards towards the new file. 
Review URL: http://webrtc-codereview.appspot.com/213003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 12:16:43 +00:00
mallinath@webrtc.org
c01c358f54 session/phone/channel.cc updates after new push of libjingle revision. 
Review URL: http://webrtc-codereview.appspot.com/225003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 09:45:24 +00:00
mallinath@webrtc.org
ebc0a00197 One of Justin comment was to have XXXXInterface and XXXX, rather than XXXX and XXXXImpl. So here are the changes, i don't like to call some the classes as interfaces like MediaStreamTrackListInterface, but they fit the criteria to be called as interface. 
Review URL: http://webrtc-codereview.appspot.com/226001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 07:04:02 +00:00
henrike@webrtc.org
03a86998cd Fixes for build errors introduced most likely earlier today. 
Review URL: http://webrtc-codereview.appspot.com/228003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 23:36:38 +00:00
wu@webrtc.org
0c378112ec Define NO_SOUND_SYSTEM for chromium build. 
Review URL: http://webrtc-codereview.appspot.com/226002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 22:35:01 +00:00
wu@webrtc.org
ebc405d9c6 Remove the fakeportallocator from the libjingle.gyp. 
Review URL: http://webrtc-codereview.appspot.com/228001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 18:36:04 +00:00
wu@webrtc.org
4ee906d297 When WEBRTC_VIDEO_ENGINE_FILE_API is not defined, disable the code in vie_file_impl.cc and vie_file_image.cc so that we can remove the libjpeg dependency. Also disable the auto test for the vie file api. 
Review URL: http://webrtc-codereview.appspot.com/227001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 17:56:38 +00:00
marpan@webrtc.org
5a3e20f678 Removed unused variables (build error) for test_fec. 
Review URL: http://webrtc-codereview.appspot.com/223001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 16:59:24 +00:00
wu@webrtc.org
6c2d7107ae * Update to use the new libjingle release. 
* Stop using any local mods for the default build (non-dev).
Review URL: http://webrtc-codereview.appspot.com/224001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 16:58:50 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine. 
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 15:19:55 +00:00
mallinath@webrtc.org
103f33b734 Changes after comments received from Justin and Harald. Few comments are not implemented like moving track implementation to base<> and then have child classes based on the type of track. 
Review URL: http://webrtc-codereview.appspot.com/217004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 14:31:20 +00:00
kjellander@webrtc.org
7951e819af Simple utility method for finding the project root dir (to be used by tests loading resource files) 
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
 2011-10-13 12:24:41 +00:00
perkj@webrtc.org
6a34d584b8 Implement MediaStreamProxy. 
This implements a proxy for MediaStreams and MediaStreamTracklists.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/217003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 08:48:43 +00:00
stefan@webrtc.org
4c059d87b3 Add metric for number of packets discarded by JB due to not being decodable 
Also fixes a couple of bugs related to sequence number wrap found while
testing.

BUG=
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Review URL: http://webrtc-codereview.appspot.com/218001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 07:35:37 +00:00
wu@webrtc.org
77d7d5455e Replace the DestroyDeviceInfo with a virtual destructor. 
Review URL: http://webrtc-codereview.appspot.com/212005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-12 16:57:53 +00:00
perkj@webrtc.org
38e400a967 Adding native client test page to test loopback. 
The test page is the same as the previouse test page but exchange offer messagesto answer messages.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-12 12:59:13 +00:00
amyfong@webrtc.org
e5542a0af5 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect) 
Fixed:
	24. Play local file (audio_long16.pcm) 
New:
	34. Record a PCM file 
	35. Play a previously recorded PCM file locally 
	36. Play a previously recorded PCM file as microphone 
Review URL: http://webrtc-codereview.appspot.com/209001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 20:30:56 +00:00