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Author SHA1 Message Date
niklase@google.com
569c805674 git-svn-id: http://webrtc.googlecode.com/svn/trunk@165 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:45:49 +00:00
niklase@google.com
9ad0cf1ae2 git-svn-id: http://webrtc.googlecode.com/svn/trunk@164 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:43:35 +00:00
niklase@google.com
f50cf1f9da git-svn-id: http://webrtc.googlecode.com/svn/trunk@163 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:33:00 +00:00
niklase@google.com
91081baf8a git-svn-id: http://webrtc.googlecode.com/svn/trunk@162 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:27:17 +00:00
niklase@google.com
aa107a635f git-svn-id: http://webrtc.googlecode.com/svn/trunk@161 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:26:47 +00:00
niklase@google.com
d784e5535c git-svn-id: http://webrtc.googlecode.com/svn/trunk@160 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:26:13 +00:00
niklase@google.com
2b774c73e2 git-svn-id: http://webrtc.googlecode.com/svn/trunk@159 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:25:57 +00:00
niklase@google.com
a207f59704 git-svn-id: http://webrtc.googlecode.com/svn/trunk@158 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:25:09 +00:00
niklase@google.com
cc63de44ae git-svn-id: http://webrtc.googlecode.com/svn/trunk@157 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:24:47 +00:00
niklase@google.com
470e71d364 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
hlundin@google.com
f0a476bf76 Add PictureID and NonReference to codec information 
The PictureID and NonReference information is now routed from the
encoder to the RTP packetizer through CodecSpecificInfo and 
RTPVideoHeaderVP8.
Review URL: http://webrtc-codereview.appspot.com/51003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-07 08:04:23 +00:00
henrika@google.com
dcdf311b6c Rollback to working sqrt version. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-07 08:05:26 +00:00
cduvivier@google.com
d0159d8eb0 aec_rdft_128: one entry point for each sign. 
Review URL: http://webrtc-codereview.appspot.com/61007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@153 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-06 23:35:37 +00:00
cduvivier@google.com
fae3b31707 Optimization/cleanup of 'aec_rfdt' initialization (constants, LUT, ...): 
* 2.7% AEC overall speedup for the straight C path.
* 3.5% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/60001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-06 18:32:59 +00:00
ajm@google.com
7c4469bf61 Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. 
Review URL: http://webrtc-codereview.appspot.com/56002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-06 17:45:37 +00:00
henrika@google.com
c5758f8c51 Uploaded test files for ADM functional tests. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-06 08:34:04 +00:00
holmer@google.com
98b4ed1ff8 Disabling DEBUG_FILE in the overuse detector by default. 
Review URL: http://webrtc-codereview.appspot.com/63001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-05 14:47:23 +00:00
tlegrand@google.com
2b4b7f1321 Moving two testfiles, audio coding module. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-05 09:17:37 +00:00
tlegrand@google.com
0adca82c35 Move iLBC test and reference files to new location. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-05 09:10:23 +00:00
henrika@google.com
435c7ce243 Fixed minor type for Mac and Linux target. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-04 16:07:30 +00:00
henrika@google.com
d6552174c4 Prepares a move all data files required by VoiceEngine into one common place. 
A new folder called test/data/voice_engine will be created and all files
will be placed here.
Review URL: http://webrtc-codereview.appspot.com/61004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-04 15:52:04 +00:00
henrika@google.com
2e8a1a2092 Creates new test folder for VoiceEngine test files and adds the required files. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-04 15:39:40 +00:00
tommi@google.com
53af7595d1 Switch the sample client back to render the videos in the main window 
instead of two popup windows.  This also demonstrates one way of
implementing the VideoRenderer interface.
Review URL: http://webrtc-codereview.appspot.com/51004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-04 12:47:37 +00:00
mikhal@google.com
cdc943e2d5 VCM: 1. Updating handling of empty packets. 2. Updating JB test. 3. Removing un-used code. 
Review URL: http://webrtc-codereview.appspot.com/59001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-01 18:15:11 +00:00
marpan@google.com
c13708271a Update media_opt_util with frame size parameters. 
Review URL: http://webrtc-codereview.appspot.com/51002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-01 17:18:53 +00:00
hlundin@google.com
6b04739e04 Route CodecSpecificInfo from encoder to packetizer 
Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-01 08:32:57 +00:00
mikhal@google.com
b5427cbd35 Changing JPEG API to to accept rawImage and encodedImage; moved video_image.h from modules/video_coding/codecs to common_video/interface, and some general re-write to JPEG, especially with regard to memory handling. Required VCM/ViE changes are also included. 
Review URL: http://webrtc-codereview.appspot.com/55002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-01 01:17:49 +00:00
marpan@google.com
67d7282900 Allow the FEC to protect up to maximum #packets (48) if the 
media packet list is above this max.
Review URL: http://webrtc-codereview.appspot.com/45005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-30 20:14:15 +00:00
cduvivier@google.com
9d94116697 Optimization of 'rftbsub': 
* scalar optimization, vectorization.
* 0.5% AEC overall speedup for the straight C path.
* 2.8% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/48008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-30 19:19:37 +00:00
leozwang@google.com
8ec2231979 Add aec_rdft.c to android build 
Review URL: http://webrtc-codereview.appspot.com/58001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-30 18:34:09 +00:00
tommi@google.com
b2e56b9816 Switch use of wsprintfW out for the libjingle equivalent. 
Review URL: http://webrtc-codereview.appspot.com/55001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-30 07:30:13 +00:00
cduvivier@google.com
20cb6b684b Optimization of 'rftfsub': 
* scalar optimization, vectorization (including new file for SSE2 code
  and path selection mechanism).
* 0.5% AEC overall speedup for the straight C path.
* 3.0% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/46005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-30 01:22:19 +00:00
leozwang@google.com
190d0873b0 Remove included header files on that unit_test is not dependent, correct error in last CL 
Review URL: http://webrtc-codereview.appspot.com/57001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-29 22:45:59 +00:00
leozwang@google.com
6fb5d19289 Add Android.mk for apm unit test and make it compile on android 
Review URL: http://webrtc-codereview.appspot.com/54001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-29 22:01:00 +00:00
mikhal@google.com
21a4405d01 VPLIB/Interpolation - Delete decode buffer only if too small, this required an API change. In addition, done some clean up and updated test and related code in VCM. 
Review URL: http://webrtc-codereview.appspot.com/46003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-29 17:00:03 +00:00
marpan@google.com
1eccf7dfb3 Some code cleanup for rtp_sender_video.cc. 
Review URL: http://webrtc-codereview.appspot.com/44003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-28 23:10:33 +00:00
tommi@google.com
8c8ef22db1 Add an owners file for the peerconnection folder. 
Review URL: http://webrtc-codereview.appspot.com/52003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-28 11:21:07 +00:00
marpan@google.com
e02b57e397 Updates to qm_select: Function to update content state, and function for FEC rate adjustment. 
Added packetLoss parameter to qm_select, and some code clean-up.
Review URL: http://webrtc-codereview.appspot.com/44009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-28 00:02:51 +00:00
leozwang@google.com
6cc3f000fc Include forward_error_correction_internal.cc which was added in #93 to android build 
Review URL: http://webrtc-codereview.appspot.com/53001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-27 16:27:18 +00:00
cduvivier@google.com
181f543de4 AEC specific version of " Real Discrete Fourier Transform". 
Lots of AEC CPU usage is coming from calls to 'rdft'. To optimize this,
deep changes (modification of memory layout, ...) have to be done and it
is not practical to do them in an utility library. Most of these changes
will occur in subsequent CLs.

The new file 'aec_core_rdft.c' is a copy of 'modules/audio_processing/
utility/fft4g.c' whose size has been significantly reduced by removing
all code non-necessary to compute rdft. The main entry point and utility
functions have also been modified to take into account the fact that all
'rdft' calls performed by AEC have a length of 128. This yields:
* 1.8% AEC overall speedup for the straight C path.
* 2.3% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/44008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-24 18:22:47 +00:00
marpan@google.com
3ad9c18843 Update on content metrics: 
Added metrics averaged over intervals of the loss/bandwidth reports, to be used for adjustment of robustness settings. Separated this set
from the (global) metrics used for resolution adaptation.
Some code cleanup in content_metrics.cc/.h.
Review URL: http://webrtc-codereview.appspot.com/52002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-24 18:08:33 +00:00
hellner@google.com
d4d27675a8 Minor update that fixes crash in system wrappers unittest. (the crash was in the test of map_wrapper). 
Review URL: http://webrtc-codereview.appspot.com/45007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 17:30:17 +00:00
marpan@google.com
0d7e5bc712 Fix bug on key frame boost allocation, and some update/cleanup to same function. 
Review URL: http://webrtc-codereview.appspot.com/50001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 16:36:33 +00:00
hellner@google.com
3c45dfd178 Fixes valgrind warnings in the rtp_rtcp module. 
Review URL: http://webrtc-codereview.appspot.com/47005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 16:24:03 +00:00
ajm@google.com
95fa29ec96 Creating a new directory for test data files, and moving audio_processing files there. 
Review URL: http://webrtc-codereview.appspot.com/48004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 11:45:12 +00:00
henrika@google.com
4bf9c0b123 Adds sanity checks related to IAudioCaptureClient::GetBuffer. 
Review URL: http://webrtc-codereview.appspot.com/45006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 09:44:59 +00:00
mflodman@google.com
a591d3fb41 Correcting two auto test errors. 
1. See diff for _main.cc.
2. Can't chose custom ports, see line 843 and 997 in new version of _custom_call.cc.

+ Refactored custom_call.cc, which makes the diff quite hard to read
Review URL: http://webrtc-codereview.appspot.com/47004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 08:43:37 +00:00
ronghuawu@google.com
36d93504b8 Remove the full header file path to: 
1) align with all the other webrtc header files.
2) and for the case(libjingle) when we want to deliver webrtc as lib and headers - all the headers will be in one folder.
Review URL: http://webrtc-codereview.appspot.com/44007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-22 21:17:43 +00:00
mikhal@google.com
2b83acef3e VCM/JB: Setting only non-empty frames for decoding (when not waiting for NACK). 
Review URL: http://webrtc-codereview.appspot.com/49001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-22 17:25:06 +00:00
tlegrand@google.com
5b95bcd22c Critical section in constructor, audio coding module 
Two changes in this CL:
-Removal of a critical section lock in the constructor of audio coding module
-Removal of one unused variable
Review URL: http://webrtc-codereview.appspot.com/43001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-22 09:21:51 +00:00