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Author SHA1 Message Date
andrew@webrtc.org
f1a605cad6 Update DEPS to support Mac clang build. 
Review URL: http://webrtc-codereview.appspot.com/244003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 15:29:16 +00:00
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module. 
BUG=
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Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 13:42:50 +00:00
stefan@webrtc.org
158f496030 Fixes a rate control bug in the VP8 wrapper. 
Changes how we signal frame rate and frame durations to the encoder. Rather
than changing the time base, we now only modify the frame durations, while
keeping the timebase constant. The frame duration is currently calculated
from the average input frame rate. Ideally, the frame duration should
be calculated as the timestamp diff, which is the real duration of a
frame, but the encoder doesn't seem to like too varying durations.

BUG=
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Review URL: http://webrtc-codereview.appspot.com/247001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 13:15:16 +00:00
perkj@webrtc.org
aa32319046 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. 
This cl also change MediaStreamProxy to only allow setting the state from the signaling thread.

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Review URL: http://webrtc-codereview.appspot.com/237001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 09:32:38 +00:00
mallinath@webrtc.org
ca8b3a387e kind() method in track interface is changed to std::string to keep uniformity with other get methods 
Review URL: http://webrtc-codereview.appspot.com/242003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 09:18:25 +00:00
mallinath@webrtc.org
96ba19034c ref_count.h file name changed to refcount.h to keep as other ( most ) files are named in libjingle. 
Review URL: http://webrtc-codereview.appspot.com/240008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 08:01:11 +00:00
stefan@webrtc.org
ead87b5051 Fix potential issue where frame buffers might be freed while being decoded. 
BUG=
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Review URL: http://webrtc-codereview.appspot.com/243004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 06:46:37 +00:00
stefan@webrtc.org
2b0f094c8f Avoid reallocating the decodedImage for every decoded frame. 
Also made sure the right size is allocated.

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TEST=

Review URL: http://webrtc-codereview.appspot.com/240004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 06:39:03 +00:00
mikhal@webrtc.org
ee3dfa6f43 Review URL: http://webrtc-codereview.appspot.com/241007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 00:46:09 +00:00
mikhal@webrtc.org
1af915d8ae video_coding: vp8: Updating error propagation threshold 
Review URL: http://webrtc-codereview.appspot.com/246002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 18:19:18 +00:00
pwestin@webrtc.org
11330b003e Added myself to rtp module watch 
Review URL: http://webrtc-codereview.appspot.com/243003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 17:54:20 +00:00
kma@webrtc.org
d75889e2eb Change of Android makefiles to build latest video coding code. 
Review URL: http://webrtc-codereview.appspot.com/239008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 16:28:56 +00:00
henrika@webrtc.org
7cf893743a git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-10-20 12:30:35 +00:00
henrika@webrtc.org
cedbb036d1 [Issue 101] Solves memory leak on Windows 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@784 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 12:11:45 +00:00
perkj@webrtc.org
2ebc9ce5a3 Fix broken PeerConnection Dev build. 
Fix MediaStreamHandler::CommitLocalStreams refactoring error.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/243005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 11:52:31 +00:00
stefan@webrtc.org
c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/241006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 08:19:34 +00:00
mallinath@webrtc.org
f553ec70c7 Notifier and RefCount interface and implementation class name changed according to the naming convention. 
Review URL: http://webrtc-codereview.appspot.com/241003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 06:24:24 +00:00
mflodman@webrtc.org
ae499a2ac8 Set correct codec info before sending frame to VCM. 
Review URL: http://webrtc-codereview.appspot.com/240003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 05:55:46 +00:00
kjellander@webrtc.org
81f25f9ff8 Fixing build errors on Windows platform. Minor changes... 
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 20:06:56 +00:00
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.

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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
 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.

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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.

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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.

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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