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Author SHA1 Message Date
henrik.lundin@webrtc.org
895870b68f Adding marker bit to RTPanalyze results 
Review URL: http://webrtc-codereview.appspot.com/254005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 08:44:42 +00:00
mikhal@webrtc.org
bb8dfbdee2 updating vpm unit_test following r858 
Review URL: http://webrtc-codereview.appspot.com/255005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 22:07:16 +00:00
turaj@webrtc.org
7395d3d8e9 Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115 
Review URL: http://webrtc-codereview.appspot.com/261002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:33:06 +00:00
turaj@webrtc.org
fac5316856 Address the problem that iSAC could not go 16 kHz. It was addressed in P4 but not moved to svn. 
Review URL: http://webrtc-codereview.appspot.com/261001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:32:46 +00:00
turaj@webrtc.org
9116cf7c9b Have a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error. 
Review URL: http://webrtc-codereview.appspot.com/239016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:29:34 +00:00
mflodman@webrtc.org
29d75b3f7d Only allow increasing capture time. 
Review URL: http://webrtc-codereview.appspot.com/259001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:10:49 +00:00
andrew@webrtc.org
18ee6ec8e9 Use __inline in NS-fixed. 
The use of "inline" was failing to build on Windows.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/255003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:07:46 +00:00
andrew@webrtc.org
3119ecfec8 Fix audioproc build errors on Windows. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/254003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:00:18 +00:00
mikhal@webrtc.org
c4ab8706f4 video_processing: Adding logic to avoid a memcpy when not required 
Review URL: http://webrtc-codereview.appspot.com/255002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 16:44:24 +00:00
punyabrata@webrtc.org
0ab521f754 Resolving a crash related to strncopy followed by a strcat 
call. strncopy will not explicity copy or add a "\0" therefore
strcat did not know where to append the "\n" which was causing
an out of bounds crash.
Because we are checking the length, strcpy should be good enough
as it also copies the "\0". Please note that that I am pre-emptively
adding 2 instead of 1 to the length to take into account of the \n
that will be added later.
Review URL: http://webrtc-codereview.appspot.com/253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 15:19:44 +00:00
kjellander@webrtc.org
d6837709cf Fixing VPMUnitTest compilation error on Windows. 
It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.

Review URL: http://webrtc-codereview.appspot.com/257002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 01:51:10 +00:00
henrike@webrtc.org
b37c628ae4 Fixes crash due to r841. 
Review URL: http://webrtc-codereview.appspot.com/256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 23:53:04 +00:00
kma@webrtc.org
e9f909b575 Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it. 
Review URL: http://webrtc-codereview.appspot.com/244002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 22:24:57 +00:00
kma@webrtc.org
27957508a3 Changed Android makefile to make the lastest video render code run. 
Review URL: http://webrtc-codereview.appspot.com/247005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:29:50 +00:00
henrike@webrtc.org
066f9e5a2f Ray, please verify that this cl fixes the issue. Once the verification has been made, please review: 
Henrik A: VoE
Andrew: audio_conference_mixer

Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 23:15:47 +00:00
henrike@webrtc.org
731ecba47d Review URL: http://webrtc-codereview.appspot.com/251002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 22:49:24 +00:00
braveyao@webrtc.org
1f6d740571 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). 
It could happen that if you want to restart playout, the new sponsored Render thread would catch this event
if the previous Render thread quits before this event is set.
With this modification, the device plugging out/in during talking would be supported well.
Review URL: http://webrtc-codereview.appspot.com/248002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 21:30:30 +00:00
stefan@webrtc.org
f960211f8b Fixes two jitter buffer bugs related to NACK. 
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 16:00:49 +00:00
stefan@webrtc.org
eb65860720 Reverts the workaround in r823 and solves a macro bug. 
The macro bug caused frames to be dropped after being grabbed
for decoding.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 12:25:34 +00:00
tina.legrand@webrtc.org
8b1f621e3a Updated gypi for tests to work on osx. 
Review URL: http://webrtc-codereview.appspot.com/250002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 08:57:34 +00:00
mikhal@webrtc.org
5200a05500 video_coding/jitter_buffer Updating condition on which we return a frame. 
Review URL: http://webrtc-codereview.appspot.com/240011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:54:54 +00:00
mikhal@webrtc.org
30f6376802 VP8: Updating codec version: VP8 version will now return the libvpx version used. 
Review URL: http://webrtc-codereview.appspot.com/247009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:45:00 +00:00
stefan@webrtc.org
2d28aff785 Workaround for an issue where frames are grabbed for decoding prematurely. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:13:18 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate. 
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:08:29 +00:00
mflodman@webrtc.org
7e4269e9ee Changed VP8 qp min and added noise reduction. 
Review URL: http://webrtc-codereview.appspot.com/248003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 12:59:47 +00:00
kjellander@webrtc.org
6b7799021c 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@819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 02:38:09 +00:00
andrew@webrtc.org
cb18121990 Add an unpacker tool for audioproc debug files. 
- It only unpacks audio data at the moment.
- Also switch to Chrome's protoc.gypi for protobuf targets. This reduces
  the complexity of our targets.

Review URL: http://webrtc-codereview.appspot.com/241009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 00:27:17 +00:00
frkoenig@google.com
fc9bcef8c7 Data alignment fix for SSIM. 
WebRtc_UWord64[2] wasn't always aligned to 128 bytes, which
is necessary for _mm_store_si128.  By declaring the 
variable as __m128i it will always be 128 bytes aligned.

Incorrect include files.

__m128i is defined in emmintrin.h for visual studio.  Extra include on mac and linux is not a problem.
Review URL: http://webrtc-codereview.appspot.com/239013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 00:07:32 +00:00
stefan@webrtc.org
d855c1a4e8 Reverts r807 and fixes the real issue in the VCM. 
This fixes an issue in the VCM where we don't wait for a packet to arrive
if the jitter buffer is empty. This also fixes an issue where an old
packet can trigger a packet event signal for a future frame.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 11:52:48 +00:00
henrika@webrtc.org
bdb55c806f This CL is an attempt to remove a crash we can see when closing down VoiceEgine. 
It can happen that the capture thread tries to access an invalid object after StopPlayout has been called.

I have also extended the usage of the new ScopedCOMInitializer to all threads. See this step as code cleanup.
Review URL: http://webrtc-codereview.appspot.com/239014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 11:03:28 +00:00
henrika@webrtc.org
a6c23357c0 Solves crash in ADM API unit test for Core Audio on Windows 
Review URL: http://webrtc-codereview.appspot.com/244009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 08:31:33 +00:00
henrika@webrtc.org
5423bc2d0b Adds correct absolute paths to all input files in ADM functional unit tests. 
Files are now read and played out correctly.
Review URL: http://webrtc-codereview.appspot.com/246006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 08:24:20 +00:00
kma@webrtc.org
ca325ececd Corrected a linux build error introduced in issue 246005. 
Review URL: http://webrtc-codereview.appspot.com/246008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 02:36:09 +00:00
wjia@webrtc.org
f0cd394a2e Put fwrite calls under corresponding macros since they shouldn't show up in release build. 
This also make chromeos build happy.
BUG=none
TEST=compile
Review URL: http://webrtc-codereview.appspot.com/247006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:40:43 +00:00
mikhal@webrtc.org
f31826e17b adding a wait on the decode thread when no frames are available 
Review URL: http://webrtc-codereview.appspot.com/246009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:20:54 +00:00
mikhal@webrtc.org
a412924c0e VP8:Setting number of cores based on image size 
Review URL: http://webrtc-codereview.appspot.com/242010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:02:43 +00:00
kma@webrtc.org
913644b92d For commiting changes in CL 277002, due to file structure changes introduced during 
the review of the code.
Review URL: http://webrtc-codereview.appspot.com/246005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 21:36:33 +00:00
andrew@webrtc.org
537096a5c1 Remove unnecessary objective-c compiler flags. 
Review URL: http://webrtc-codereview.appspot.com/239011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 15:07:08 +00:00
henrika@webrtc.org
bf478faebb Ensures that ADM unit tests builds on all platforms. 
Review URL: http://webrtc-codereview.appspot.com/240009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 10:31:02 +00:00
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module. 
BUG=
TEST=

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=
TEST=

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
stefan@webrtc.org
ead87b5051 Fix potential issue where frame buffers might be freed while being decoded. 
BUG=
TEST=

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.

BUG=
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
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. 
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
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