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Author SHA1 Message Date
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
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
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
stefan@webrtc.org
c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774. 
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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
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
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=
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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
 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
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
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
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
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
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
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
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
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
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.

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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
wu@webrtc.org
ea89922b56 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. 
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Review URL: http://webrtc-codereview.appspot.com/213002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 17:13:51 +00:00
andrew@webrtc.org
199f4defd3 Rename all .cc files which include Objective-C headers to .mm. 
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
 2011-10-11 15:43:35 +00:00
stefan@webrtc.org
791eec7424 Add API to get the number of packets discarded by the video jitter buffer due to being too late. 
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Review URL: http://webrtc-codereview.appspot.com/200001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 07:53:43 +00:00
stefan@webrtc.org
06887aebae Fixes two bugs when decoding with packet losses. 
Disable _missingFrame bit since we can't set it correctly with FEC.

No longer return more than one decoded frame per Decode() call.
This is a work-around for a bug where the frame info map was popped more often than items were added to the map.

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Review URL: http://webrtc-codereview.appspot.com/215001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 14:17:46 +00:00
kjellander@webrtc.org
25e0b8e3a0 Python output flag and keyframe interval flags. 
Refactored main method into using 6 helper methods for better overview.

Review URL: http://webrtc-codereview.appspot.com/207001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-07 07:52:00 +00:00
kjellander@webrtc.org
a31b254084 Python output flag and keyframe interval flags. 
Refactored main method into using 6 helper methods for better overview.

Review URL: http://webrtc-codereview.appspot.com/207001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-07 06:50:22 +00:00
henrike@webrtc.org
bf54ef9bb7 Removed code under a non-existing define. 
Review URL: http://webrtc-codereview.appspot.com/193006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 18:14:25 +00:00
andrew@webrtc.org
b2d4921f3b Remove trailing whitespace in AudioDevice. 
(That I introduced...)
Review URL: http://webrtc-codereview.appspot.com/198002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 16:34:36 +00:00
kjellander@webrtc.org
35a1756502 First version of video quality measurement program and test framework. 
See https://docs.google.com/a/google.com/document/d/1w6Nrxw6yTg_sDu18Ux8oZPEMo5F_R-zt62udrmmTeOc/edit?hl=en_US
for background, details and additional instructions on usage.

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Review URL: http://webrtc-codereview.appspot.com/175001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 06:44:54 +00:00
kma@webrtc.org
af57de006a Some code style changes in audio_processing/ns/main/source/ by Astyle, 
with a little manual modification.
Review URL: http://webrtc-codereview.appspot.com/201002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 23:36:01 +00:00
henrik.lundin@webrtc.org
01ca01f6e6 Adding neteq_tests to modules tests 
Also moving neteq_tests.gyp and renaming to gypi. Cleaning up a
little in neteq_tests.gypi.

Review URL: http://webrtc-codereview.appspot.com/191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 20:38:19 +00:00
kma@webrtc.org
bbc1f10187 Changed modules/audio_processing/utility/Android.mk, to correct a build error in 
Android with the change from version r674.
Review URL: http://webrtc-codereview.appspot.com/197003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 18:09:02 +00:00
kma@webrtc.org
bf39ff4271 Some general optimization in NS. 
No big effort in introducing new style.
Speed improved ~2%.
Bit exact.
Will introduce mulpty-and-accumulate and sqrt_floor next, which increase speed another 2% or so.

Note: In function WebRtcNsx_DataAnalysis, did the block separation because I found one "if" case is more frequent than "else" within a for loop; rest is kind of code re-aligning.
Review URL: http://webrtc-codereview.appspot.com/181002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 17:10:06 +00:00
stefan@webrtc.org
4b6f747373 Fixes a newly introduced bug in the jitter buffer where buffer reallocation 
causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f Fixed bug in jitter buffer which caused the missingFrames bit to never be set. 
Also updated the VP8 wrapper to return fully concealed frames (for rendering).

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Review URL: http://webrtc-codereview.appspot.com/190003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:48:11 +00:00
stefan@webrtc.org
61b4abf1f8 Proper use of frame rate argument in generic_codec_test. 
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Review URL: http://webrtc-codereview.appspot.com/181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:40:21 +00:00
mikhal@webrtc.org
e06be4f678 video coding tests: Adding ssimFrame to interface 
Review URL: http://webrtc-codereview.appspot.com/188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:43 +00:00
mikhal@webrtc.org
ae7a0522c5 video_coding robustness: Updating hybrid mode's settings 
1. Disabling adjustment factor - temporary update. 
2. Enabling a windowed filtered loss for the hybrid mode.  
Review URL: http://webrtc-codereview.appspot.com/192003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:34 +00:00
marpan@google.com
f1f3fb33b5 Update to rate-mismatch factor in media_opt_util. 
Review URL: http://webrtc-codereview.appspot.com/193003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 19:09:45 +00:00
stefan@webrtc.org
5b91464edf Allow an aggregated partition to spill over to a new packet. 
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

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Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 10:26:12 +00:00
bjornv@google.com
1ba3dbecbb Adds possibility to log delay estimates in AEC. 
Review URL: http://webrtc-codereview.appspot.com/178001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 08:18:10 +00:00
kma@google.com
c611b1a950 Bit-exact with non-Neon version. 
Review URL: http://webrtc-codereview.appspot.com/180002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 16:03:38 +00:00
andrew@webrtc.org
18421f2063 Remove unnecessary include from NS interface. 
http://code.google.com/p/webrtc/issues/detail?id=46
Review URL: http://webrtc-codereview.appspot.com/183001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:50:52 +00:00