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965 Commits

Author SHA1 Message Date
henrike@webrtc.org
7136990a3f Removed usage of the deprecated critical section constructor in udp_transport. 
Review URL: http://webrtc-codereview.appspot.com/321005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 19:17:28 +00:00
andrew@webrtc.org
986fab1496 Clean up file wrapper a bit further. 
- Make error handling in Read, Write and WriteText consistent.
- Improve the interface comments a bit.

TEST=voe_auto_test, vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/321012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 19:11:41 +00:00
leozwang@webrtc.org
0c839fe873 Add new source file to makefile 
Review URL: http://webrtc-codereview.appspot.com/322015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 19:10:24 +00:00
henrike@webrtc.org
bfa80ce95e Removed usage of the deprecated critical section constructor in system_wrappers. 
Review URL: http://webrtc-codereview.appspot.com/322004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 17:59:58 +00:00
henrik.lundin@webrtc.org
0a10e3c4b2 Fix order of include and guard in tick_time_interface.h 
Review URL: http://webrtc-codereview.appspot.com/331002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 16:08:36 +00:00
mflodman@webrtc.org
091029ba26 Refactored ViEFileRecorder. 
Types and arguments will be done in a  later CL.

Review URL: http://webrtc-codereview.appspot.com/317008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 15:31:47 +00:00
mflodman@webrtc.org
03c06505fb Refactored ViEChannel. 
Pointers/references and types will be in a future CL.

Review URL: http://webrtc-codereview.appspot.com/322016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1205 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 15:23:08 +00:00
henrik.lundin@webrtc.org
c74b2861f3 Fix the include in fake_tick_timer_interface.h 
The include was in error.

Review URL: http://webrtc-codereview.appspot.com/330002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 11:28:31 +00:00
phoglund@webrtc.org
610e90e910 Completed rewrite of codec test. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 10:40:19 +00:00
mflodman@webrtc.org
e8be22c192 Refactored ViEChannelManager ViEInputManager. 
Pointers/references and types will come in a future CL.

Review URL: http://webrtc-codereview.appspot.com/317012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 10:19:29 +00:00
leozwang@webrtc.org
e0e07bbaa0 Change file name because of r1199 
Review URL: http://webrtc-codereview.appspot.com/320013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 00:51:45 +00:00
kma@webrtc.org
ee36b9587d corrected android makefile for isac build. 
Review URL: http://webrtc-codereview.appspot.com/321013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 00:18:45 +00:00
andrew@webrtc.org
59ccd5c71f Rename _windows.h -> _win.h in system_wrappers. 
- Also rename _dummy -> no_op which states its purpose more clearly.
- Always use exclusion lists (i.e. sources! instead of sources)

TEST=builds and passes system_wrapper_unittest on Linux, Mac, Win

Review URL: http://webrtc-codereview.appspot.com/317007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 00:17:43 +00:00
kma@webrtc.org
6a17340db5 Review URL: http://webrtc-codereview.appspot.com/318014 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 22:16:57 +00:00
leozwang@webrtc.org
5fddbeb7e5 Build libyuv for webrtc 
Review URL: http://webrtc-codereview.appspot.com/322012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 20:03:26 +00:00
leozwang@webrtc.org
eda2da796e Fix compilation errors 
Review URL: http://webrtc-codereview.appspot.com/322014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 20:03:09 +00:00
kma@webrtc.org
a30093bb85 Added one file associated with check in in r1192. 
Review URL: http://webrtc-codereview.appspot.com/320012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 19:57:23 +00:00
leozwang@webrtc.org
9aa9f44ebc Add new source files because of r1174 
Review URL: http://webrtc-codereview.appspot.com/320011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 19:08:33 +00:00
kma@webrtc.org
f0a964dc0a Optimized WebRtcIsacfix_NormLatticeFilterMa() function for iSAC fix for ARM Neon 
architecture with intrinsics and assembly code. The total iSAC codec speech improved
about 3~5%.

Notes
(1) The Neon version after this optimization is not bit-exact with the generic
C version. The out quality, however, is not worse as verified by test vectors ouput,
and undertandably in theory (32bit x 32bit in Neon is more accurate than the approximation
C code in the generic version).
(2) In Android, a isac neon library will be built. Along with some new function structures,
it is partly for preparation of introducing a run time detection of Neon architecture soon.
Review URL: http://webrtc-codereview.appspot.com/268016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:59:43 +00:00
mflodman@webrtc.org
02afbeaca5 Refactored ViERenderManager. 
Will follow up with a new CL for pointer/references and functino arguments.

Review URL: http://webrtc-codereview.appspot.com/323013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:50:47 +00:00
kma@webrtc.org
6601902504 Introduced WebRtcSpl_SatW32ToW16 to iSAC fix, for Android platforms. 
Review URL: http://webrtc-codereview.appspot.com/315005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:41:07 +00:00
leozwang@webrtc.org
f147bbc878 Change codec test app lib dependency from webrtc lib to codec library 
Review URL: http://webrtc-codereview.appspot.com/317009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:22:41 +00:00
andrew@webrtc.org
7e5ddf5aa3 Restore behavior to FileWrapper::Read. 
- Returning the number of bytes read was mistakenly removed in r1175 in
  an overzealous attempt to unify the interface.
- Now both Read and WriteText return the number of bytes/characters
  processed. Write unfortunately cannot be easily changed due to the
  inheritance from OutStream.
- Improve the interface comments.

TBR=henrika@webrtc.org
BUG=issue196, issue198
TEST=voe_auto_test passes at last...

Review URL: http://webrtc-codereview.appspot.com/326001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:02:02 +00:00
henrike@webrtc.org
7cdcde3460 Removed usage of the deprecated critical section constructor in media_file. 
Review URL: http://webrtc-codereview.appspot.com/321004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 17:27:58 +00:00
stefan@webrtc.org
780a07a843 Fix infinite loop bug introduced in r1174. 
Merges CleanUpSizeZeroFrames with CleanUpOldFrames, and changes the
behavior to go through all frames looking for empty frames.

TBR=mikhals

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 15:55:19 +00:00
pwestin@webrtc.org
9fe3d51372 Set the new layer sync bit in the VP8 info struct. 
Review URL: http://webrtc-codereview.appspot.com/324010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 15:13:04 +00:00
phoglund@webrtc.org
667eca6290 Rewrote the hardware-before-streaming test. 
Restructured the test hierarchy somewhat - there is now a fixture for before-voe-init time and one for after-voe-init time.

Rewrote the hardware-before-streaming test.
Separated unrelated tests out from the rtp_rtcp tests.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 13:55:34 +00:00
henrik.lundin@webrtc.org
fbf5af444b Adding a mockable wrapper class for TickTime in VCM 
The class is called TickTimeInterface, with a fake implementation in FakeTickTime.

Review URL: http://webrtc-codereview.appspot.com/323012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 10:36:10 +00:00
stefan@webrtc.org
ef5247b5b1 Fix session_info_unittest error. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 10:25:38 +00:00
stefan@webrtc.org
0c40d3315f Fixes an assert triggered in jitter_buffer_test and disables deblocking. 
When deblocking is enabled the first frames can include uninitialized
memory. Disabling for now.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 09:39:30 +00:00
mflodman@webrtc.org
7991c0501f Refactor ViEFilePlayer. 
Types and arguments will be done in a  later CL.

Review URL: http://webrtc-codereview.appspot.com/324002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1180 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 08:38:37 +00:00
mflodman@webrtc.org
e6f64835a0 Refactored ViECapturer. 
Types and function arguments will come in a later CL.

Review URL: http://webrtc-codereview.appspot.com/322011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 08:36:01 +00:00
mflodman@webrtc.org
9a8fa4e65d Refactored vie_manager_base.*. 
The other files are only due to inheritance and will be refactored later. Same goes for pointer, references and function arguments.

Review URL: http://webrtc-codereview.appspot.com/318003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 08:18:42 +00:00
andrew@webrtc.org
6d609b59f3 Fix crashes due to static_instance. 
- Initialize a needed critsect in the constructor of
  UdpSocket2ManagerWindows.
- Don't return NULL when creating a static instance.

TEST=voe_auto_test on Windows.

Review URL: http://webrtc-codereview.appspot.com/324008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 02:36:33 +00:00
andrew@webrtc.org
5a9c6f26ab Fix max size and read-only errors in Write(). 
- A size of zero is now correctly interpreted as unlimited.
- The read-only flag is correctly checked.

TBR=henrika@webrtc.org
TEST=vie_auto_test (for real this time...)

Review URL: http://webrtc-codereview.appspot.com/315007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 00:53:30 +00:00
andrew@webrtc.org
5ae19de3ec Fix error in RtpDump::Start due to r1156. 
- r1156 fixed a check on the _text member of FileWrapper. Turns out this
  was incompatibile with the RTP dumps, which want to write both binary
  and text data. Writing text data to a file open as "b" isn't actually
  an error, so I simply removed the check.
- Also cleans up the interface, most notably removing all WebRtc types.

TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/317005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 22:59:33 +00:00
mikhal@webrtc.org
832cacacff video-coding: Adding a decoded state to the JB logic (JB refactor). 
This new class stores the last decoded info, including temporal info. 
Review URL: http://webrtc-codereview.appspot.com/300005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 21:15:05 +00:00
henrike@webrtc.org
65573f2922 Removed usage of the deprecated critical section constructor in rtp_rtcp. 
Review URL: http://webrtc-codereview.appspot.com/315004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 19:17:27 +00:00
stefan@webrtc.org
f4c8286222 Pass NACK and FEC overhead rates through the ProtectionCallback to VCM. 
These overhead rates are used by the VCM to compensate the source
coding rate for NACK and FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 15:38:14 +00:00
henrik.lundin@webrtc.org
1ced840893 Fixing a nit in the unittest 
This caused some of the build bots to fail.

Review URL: http://webrtc-codereview.appspot.com/324005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 14:59:00 +00:00
henrik.lundin@webrtc.org
eda86dc76b Adding a LayerSync bit to VP8 RTP header 
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.

The new bit is yet to be used by the VP8 wrapper.

Review URL: http://webrtc-codereview.appspot.com/323008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 14:11:06 +00:00
henrik.lundin@webrtc.org
4aae0e489f Shaping up formatting of rtp_utility_test.cc 
Preparations for future work in this file.

Review URL: http://webrtc-codereview.appspot.com/318011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 09:15:21 +00:00
bjornv@webrtc.org
0edb25dcc9 Removed valgrind warnings in resampler_unittest. 
Valgrind complained on uninitialized values in resampler_unittest. Added initialization of the member variable data_in_ in the tests.
Review URL: http://webrtc-codereview.appspot.com/322006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 09:06:54 +00:00
stefan@webrtc.org
076fa6e674 The second step towards a list based SessionInfo. 
Added unittests for most of public functions of SessionInfo.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 07:54:56 +00:00
wjia@webrtc.org
c28e7980ef exclude trace_windows.cc and trace_posix.cc when building with Chromium. 
BUG=none
TEST=compiles
Review URL: http://webrtc-codereview.appspot.com/324004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 02:16:11 +00:00
mikhal@webrtc.org
71d6391716 libyuv: fixing a bug in RotateI420 and updating test 
Review URL: http://webrtc-codereview.appspot.com/324003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 00:56:11 +00:00
mikhal@webrtc.org
352ade7023 video_coding: Allocating encoded buffer based on length and not size 
Review URL: http://webrtc-codereview.appspot.com/318010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 00:54:28 +00:00
phoglund@webrtc.org
fe61bc3607 Merge branch 'master' into voe_create_test 
Fixed broken build.

Nit fix.

Fixed style issues.

Removed accidental comment-out.

Removed test that no longer makes sense.

Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 17:02:16 +00:00
phoglund@webrtc.org
6418a24795 Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/322003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 16:24:23 +00:00
stefan@webrtc.org
1480f02faf Fix VCM test build warnings on Mac with clang. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 13:45:59 +00:00
stefan@webrtc.org
7889a9b49a Remove use of CriticalSectionScoped(CriticalSectionWrapper& critsect) in VCM. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 08:18:24 +00:00
mikhal@webrtc.org
ea71440aec video_coding: Adding the non reference flag to the receive side logic. 
Review URL: http://webrtc-codereview.appspot.com/323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 02:29:34 +00:00
andrew@webrtc.org
114c790be7 Remove character limit in WriteText(). 
- vfprintf can be used directly here, removing the need for the interim
  buffer. This change allows us to remove the artificial character limit.
- Fix bugs with _text. It wasn't actually getting set earlier, and the
  check was wrong.
- Remove asserts that should use real error checks.

TEST=DataLog and VoECallReport (through voe_auto_test), the only users of WriteText().

Review URL: http://webrtc-codereview.appspot.com/323001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-10 02:33:33 +00:00
henrike@webrtc.org
2f47b5a70f Fixes a build error when disabling trace (which is done when building with chrome flag is set). 
Review URL: http://webrtc-codereview.appspot.com/318006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-10 00:44:47 +00:00
wjia@webrtc.org
c6b286fc04 add correct include paths for both chrome build and standalone build. 
BUG=none
TEST=compiles
Review URL: http://webrtc-codereview.appspot.com/320008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-10 00:20:54 +00:00
andrew@webrtc.org
bbea716117 Workaround for libyuv libjingle breakage. 
libjingle depends on ConvertFromI420. This was previously available
through vplib. libjingle still has access to the vplib header, but the
implementation is no longer built.

Fortunately, the libyuv wrapper can supply the implementation, if we
hack the signature to return to the unsigned int types. We'll remove
this once libjingle has been updated to use libyuv directly.

Also, roll libyuv to r100 which fixes a gyp warning on Windows.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 19:43:12 +00:00
henrike@webrtc.org
315282c01a Fixes a compiler warning related to dynamically allocated static memory. the fix is to leak the memory since the OS will clean it up anyways. This will not add noise to memory tools so it's ok. The issue is reported here: http://code.google.com/p/webrtc/issues/detail?id=147. 
Review URL: http://webrtc-codereview.appspot.com/267023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 17:46:20 +00:00
mflodman@webrtc.org
d5651b98c5 Refactored ViEFrameProviderBase. 
Only style changes, ointers/references and functions will come in a later CL.

vie_capturer.cc and vie_file_player.cc are only changed du to inheriting protected members from ViEFrameProviderBase.

Review URL: http://webrtc-codereview.appspot.com/324001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 15:20:58 +00:00
xians@webrtc.org
0744ee563d Disable API tests on ALSA since the tests don't work for all the alsa devices. 
Review URL: http://webrtc-codereview.appspot.com/317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 14:05:29 +00:00
henrik.lundin@webrtc.org
6198624815 Remove warnings on Mac (Issue 178) 
Remove an if-else that can never execute the else statement.
Remove double parenthesis.

BUG=http://code.google.com/p/webrtc/issues/detail?id=178
TEST=

Review URL: http://webrtc-codereview.appspot.com/318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 13:58:17 +00:00
mflodman@webrtc.org
5cc4dc9e0c Remove warnings in VideoEngine, capture module and render module. 
BUG=164, 176, 180

Review URL: http://webrtc-codereview.appspot.com/303004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 10:12:57 +00:00
mflodman@webrtc.org
b19582b7dc Add pointer constructor to CriticalSectionScoped. 
Mainly added to simplyfy the code, e.g. when having critsect as scoped_ptr in classes.

Review URL: http://webrtc-codereview.appspot.com/302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 10:02:16 +00:00
henrikg@webrtc.org
af225d6bf6 The change http://webrtc-codereview.appspot.com/299001 (commit 1062) does not do what it intends (exclude codecs from Chromium build). This is a fix for that. webrtc.gyp is not pulled in Chromium, hence it has no effect putting a define there. Moving it to src/build/common.gypi. 
Review URL: http://webrtc-codereview.appspot.com/315002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 09:58:39 +00:00
mflodman@webrtc.org
5885a4162a Refactored ViERenderer. 
Only style changes, function and type changes will come in a later CL.

Review URL: http://webrtc-codereview.appspot.com/321001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 05:52:32 +00:00
mikhal@webrtc.org
2ab104e6be Switching WebRtc to LibYuv. 
General Notes:
1. In general, API structure was not modified and is based on VPLIB. 
2. Modification to API: Return values are based on libyuv, i.e. 0 if ok, a negative value in case of an error (instead of length). 
3. All scaling (inteprolation) is now done via the scale interface. Crop/Pad is not being used.
4. VPLIB was completely removed. All tests are now part of the libyuv unit test (significantly more comprehensive and based on gtest).   
5. JPEG is yet to be implemented in LibYuv and therefore existing implementation remains.
Review URL: http://webrtc-codereview.appspot.com/258001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 02:46:22 +00:00
mikhal@webrtc.org
ffa0a9e9c9 updating libyuv to latest version (98). 
This CL also includes some additional adaptations to the code due to the upgrade. 
Review URL: http://webrtc-codereview.appspot.com/306001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 22:45:53 +00:00
mallinath@webrtc.org
7766e2a82d - This issue raised by the chromium team when clang compiler is used. This was not an error as in this case we were accessing IPV6 address with IPV4 struct which is defined as 14 bytes in the header file, but we had the runtime check to determine the address space. 
Now the solution is to use IPV6 structures instead of IPV4 when address space is determined.

I haven't put the new solution behind AF_INET6 flag, as i don't think it's necessary. 
Review URL: http://webrtc-codereview.appspot.com/291014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 21:37:19 +00:00
andrew@webrtc.org
b0be7aa7ae Remove deprecated OS X Core Audio APIs. 
We no longer support the 10.4 SDK, so we can remove the weak-leaking
feature and exclusively use the added-in-10.5 APIs.

BUG=issue143
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/322001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 20:15:36 +00:00
marpan@webrtc.org
63b50f60d6 test_fec: Fix to valgrind warnings. 
Review URL: http://webrtc-codereview.appspot.com/304002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 19:05:39 +00:00
mikhal@webrtc.org
f5ee1dc3e6 video_coding: Adding temporal layer info support to receive side 
Review URL: http://webrtc-codereview.appspot.com/303005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 19:04:47 +00:00
xians@webrtc.org
832d7c6000 Disable typing detection for chromium since CGEventSourceKeyState is violating chromium sandbox. 
Review URL: http://webrtc-codereview.appspot.com/320003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 16:45:46 +00:00
phoglund@webrtc.org
dd094fd6ae Started extracting methods out of the main test. 
Started extracting methods out of the main test, which will hopefully make us able to make the tests independent.

Merge branch 'master' into voe_split_methods

Conflicts:
	src/voice_engine/main/test/auto_test/voe_extended_test.cc
	src/voice_engine/main/test/auto_test/voe_extended_test.h
	src/voice_engine/main/test/auto_test/voe_standard_test.cc
	src/voice_engine/main/test/auto_test/voe_standard_test.h

Extracted methods out of the standard test.

Added space before inheritance colons.

Rolled back some header file changes.

Fixed long lines.

Fixed long lines.

Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/313001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 15:07:59 +00:00
henrik.lundin@webrtc.org
d03718d1e4 Use ResourcePath in NetEQ unittest 
Review URL: http://webrtc-codereview.appspot.com/320001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 11:42:43 +00:00
mflodman@webrtc.org
d2ee5d989d Changed sync bug introduced in refactoring. 
Review URL: http://webrtc-codereview.appspot.com/319001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 09:25:11 +00:00
mflodman@webrtc.org
c78209c58b Add log when transport fails to send packet. 
Review URL: http://webrtc-codereview.appspot.com/311002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 09:22:45 +00:00
kjellander@webrtc.org
7de6e10410 Fixing compilation error on Linux 64-bit 
Problem was introduced in http://webrtc-codereview.appspot.com/311001/ because I had projects generated with Valgrind configuration, which is more forgiving about these implicit conversions.

BUG=
TEST=Compiling in Debug+Release on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/318002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 08:39:13 +00:00
kjellander@webrtc.org
5b97b1216f Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp. 
Fixed unit tests so they don't use ASSERT_DEATH since that doesn't work with Valgrind.

Fixed all Valgrind warnings except the one caused by CriticalSectionWrapper in system_wrappers.

Reworked all includes and GYP include paths to use full directory paths.

Removed util.h for logging, since it rendered warnings in Valgrind because of gflags. Replaced it with a verbose flag and a new function in video_quality_measurement.cc

BUG=
TEST=Passed test_support_unittests and video_codecs_test_framework_unittests on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/311001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 07:42:18 +00:00
henrike@webrtc.org
441b3fe2a1 Made some global statics have function scope so that the global static count is 0 for the rtp_rtcp module. 
Review URL: http://webrtc-codereview.appspot.com/316001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 02:03:49 +00:00
stefan@webrtc.org
cc7b649474 Add trace for the situation when the min bitrate > available bandwidth. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/312001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 13:22:06 +00:00
phoglund@webrtc.org
693240f2d9 Fixed many formatting and indentation problems in voe_auto_test. 
Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 12:32:58 +00:00
henrik.lundin@webrtc.org
598ad06432 Fixing compiler warning in NetEQ 
With some compiler settings, a warning was issued for NetEQ,
saying that pw16_randVec was accessed out of bounds.
This did never happen in practice, but this change makes the
compiler understand this.

Review URL: http://webrtc-codereview.appspot.com/309001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 11:52:09 +00:00
stefan@webrtc.org
b3bd1cd5f1 Fixes Valgrind warnings in the default VCM tests. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/299010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 11:48:09 +00:00
henrik.lundin@webrtc.org
bf86c33b0e Removing OutputDebugString from rtp_rtcp module 
This is in response to WebRTC issue 167.

BUG=http://code.google.com/p/webrtc/issues/detail?id=167

Review URL: http://webrtc-codereview.appspot.com/301013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 10:44:05 +00:00
henrik.lundin@webrtc.org
44ef3774ce Fixing a compiler error in NetEQ 
This error would only arise when compiling without support for
DTMF (which is not the default config).

Review URL: http://webrtc-codereview.appspot.com/310001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 10:43:25 +00:00
phoglund@webrtc.org
5b343aedcc Added missing .h files to .gypi files so they will show up in xcode / vc projects. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 10:34:35 +00:00
stefan@webrtc.org
58927e8d8f Disable deblocking temporarily due to Valgrind warnings. 
Also corrects the copying of the decoded image data for frames
with odd width or height.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/307002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 08:13:31 +00:00
marpan@webrtc.org
1d34212a45 FEC: Update to packets masks (FEC generator matrix) in fec_private_tables.h 
A set of the packet masks (up 10x10 size) are modified for the following reasons:

1) have more even column and row degree (number of 1 bits), when possible.

2) if cases where the column degree cannot be constant across source packets, placed the extra 1 bit in the first packet column (so little more protection on 1st partition), as opposed to having some ~middle source packet have the extra bit.

3) in some cases, made the mask a little more sparse/reduced the overlap.

Overall the average recovery is a little better with these masks.

Mask sizes above 10 will be updated in future changelist.
Review URL: http://webrtc-codereview.appspot.com/305001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 18:50:11 +00:00
kma@webrtc.org
4a8b1eaf6e In NS, replaced a divide calculatoin by shifting, and thus saved the MIPS by 5%(ARMv7) and 10%(ARMv7-Neon). Bit is not exact with the original. Quality is similar. 
Review URL: http://webrtc-codereview.appspot.com/298004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 18:04:48 +00:00
henrik.lundin@webrtc.org
b6e58eb5a1 Fix formatting of rtp_format_vp8* 
Sorting out all lint issues and fixing indentation.

Review URL: http://webrtc-codereview.appspot.com/301011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 15:56:18 +00:00
stefan@webrtc.org
c7e2bffb66 Fix header/lib mismatch caused by a constant not defined for header file. 
BUG=http://code.google.com/p/webrtc/issues/detail?id=170
TEST=

Review URL: http://webrtc-codereview.appspot.com/300008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 13:44:25 +00:00
phoglund@webrtc.org
048b037342 Fixed vie_auto_test shutdown race conditions. 
Fixed a race condition crash in vie_auto_test shutdown. Certain tests did not clean up the voice engine properly which caused crashes during certain uncommon timing conditions.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/307001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 10:42:05 +00:00
xians@webrtc.org
eff3c8905f this patch fixes the valgrind warnings in the adm api test for pulseaudio in linux. 
Review URL: http://webrtc-codereview.appspot.com/301012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 10:02:56 +00:00
mikhal@webrtc.org
cae01010bd libyuv unit test: adding check for fread return value 
Review URL: http://webrtc-codereview.appspot.com/303007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 00:44:16 +00:00
mikhal@webrtc.org
a5e980a906 Updating jitter buffer test following latest changes. 
Review URL: http://webrtc-codereview.appspot.com/294002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 18:27:31 +00:00
phoglund@webrtc.org
23e1c0a0b1 File handling in vie_auto_test now uses fileutils so input and output file end up in a good place. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 15:27:04 +00:00
perkj@webrtc.org
ec7759a8c4 Fix broken vie_capture_module_test on mac. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/303006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 12:17:10 +00:00
perkj@webrtc.org
8627adc158 Refactored Video capture Unit test to use gtest. 
Fix Valgrind warnings on Linux.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/296009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 09:58:55 +00:00
stefan@webrtc.org
0ae71b9ccb Disable temporal layers when building with Chromium. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 08:42:52 +00:00
henrika@webrtc.org
af71f0e5d9 Fixes two minor issues reported by the Coverty Integration Manager. 
BUG=none
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/302002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 07:02:22 +00:00
andrew@webrtc.org
c9cc3750cf Add missing system_wrappers dependency. 
TBR=kma@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/301009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 20:51:20 +00:00
kma@webrtc.org
b59c031660 For Android ARMv7 platforms, added a feature of dynamically detecting the existence of Neon, 
and when it's present, switch to some functions optimized for Neon at run time.
Review URL: http://webrtc-codereview.appspot.com/268002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 18:34:50 +00:00
andrew@webrtc.org
ae7017d588 Fix missing dependency in audioproc. 
TBR=bjornv@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/300006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 01:43:29 +00:00
andrew@webrtc.org
7bf2646e4d Make protobuf use optional. 
- By default, disable the AudioProcessing protobuf usage in the Chromium
  build. The standalone build is unaffected.
- Add a test for the AudioProcessing debug dumps.

TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/303003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 00:03:31 +00:00
mflodman@webrtc.org
626fbfd4cd Correcting vie_encoder nits. 
Review URL: http://webrtc-codereview.appspot.com/302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 23:39:11 +00:00
perkj@webrtc.org
6b1bfd6c5e Changed webrtc::ACMCodecDB::neteq_decoders_ to a const array. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 12:48:19 +00:00
pwestin@webrtc.org
db221d2b81 Fixes to temporal layers, Henrika please review src/common_types.h 
Review URL: http://webrtc-codereview.appspot.com/286001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 11:31:08 +00:00
phoglund@webrtc.org
6aed73d218 Fixed release compilation error. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/298003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 11:14:12 +00:00
henrik.lundin@webrtc.org
e26aad4a9e Disable NetEQ unittest for Windows 
Disable NetEqDecodingTest::TestNetworkStatistics for Windows.
It was never tested for Windows. Something is causing it to
fail, probably need different set of test vectors.

Review URL: http://webrtc-codereview.appspot.com/302003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 10:27:14 +00:00
stefan@webrtc.org
9cb2b56b65 Corrected a fread verification. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 10:22:29 +00:00
phoglund@webrtc.org
b956b4856a vie_auto_test may now be run in automated mode on all three platforms. 
Fixed chrash bug on Mac, but there are still crash bugs since a couple weeks back. These will have to be fixed separately.

Removed dialogs from capture tests on Windows.

Removed some dead code related to answer files.

Added the last Windows fixes.

Fixed the Mac vie_auto_test runner - it will now run on Mac again. It will still crash randomly on codec and rtcp tests though.

Fixed compilation error.

Got patch to commit on Mac.

Temp commit on mac

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/292011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 10:19:27 +00:00
perkj@webrtc.org
38ca4f2953 Fix code review comments. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:34:10 +00:00
perkj@webrtc.org
d3eac4158c Fixed webrtc::perm variable. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:34:01 +00:00
perkj@webrtc.org
1b72fcd27b Fix symbol RTPFILE_VERSION. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:33:51 +00:00
stefan@webrtc.org
772d70bcd2 Fix release build error. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:18:49 +00:00
stefan@webrtc.org
a4a88f90c4 Implemented NACK based reference picture selection. 
This CL implements NACK based reference picture selection for VP8. A separate
class is used for keeping track of the references and managing the VP8 encode
flags. Appropriate tests have also been added.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/284002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 08:34:05 +00:00
henrike@webrtc.org
4b00560a6e Fixes build error in rtp_rtc module introduced in r1076. 
Review URL: http://webrtc-codereview.appspot.com/301005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 00:32:24 +00:00
punyabrata@webrtc.org
c1ed87602a Adding some error handling functionality in the windows audio core implementation to 
stop rendering automatically and throw a playout-error callback when RequestPlayoutData
fails
Review URL: http://webrtc-codereview.appspot.com/300003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:55:35 +00:00
mflodman@webrtc.org
c6182915a3 Fix vie_encoder.cc. 
TBR=ajm

Review URL: http://webrtc-codereview.appspot.com/301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:45:28 +00:00
mflodman@webrtc.org
84d17838ac Refactored ViEEncoder. 
Style changes + QT Metrics class from h-file to cc-file, type changes will be in another CL.

Review URL: http://webrtc-codereview.appspot.com/303001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:02:23 +00:00
kjellander@webrtc.org
5f4f69ac57 Removing sleeps from vp8_test. 
These sleeps were remains from earlier tests that required them to work with some codecs. Removing these sleep calls cut the execution time from 90s to 30s on my machine.

Review URL: http://webrtc-codereview.appspot.com/304004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:50:04 +00:00
pwestin@webrtc.org
0644b1dc35 Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions 
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.

An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).

Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.

The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):

Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo

Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:42:31 +00:00
bjornv@webrtc.org
132feb1270 Made tables static. 
In this CL global tables have been moved to where they are actually used. If for some reason they need to be available in a larger scope we can add them again at that point.
Review URL: http://webrtc-codereview.appspot.com/303002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:40:50 +00:00
kjellander@webrtc.org
4c4b7f500f Converting vp8_test to use fileutils and gtest 
Review URL: http://webrtc-codereview.appspot.com/289012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:24:36 +00:00
tina.legrand@webrtc.org
f64162c335 Adding const to a number of constant tables. Setting some tables to static. 
Patch set 2: Renaming static const tables. They no longer need the prefix WebRtc_Isac...
Review URL: http://webrtc-codereview.appspot.com/301001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 13:01:39 +00:00
bjornv@webrtc.org
bedabb25bf Added const on const tables. 
Builds on Linux.

Tommi: Can you try on Windows?
Review URL: http://webrtc-codereview.appspot.com/300002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 07:44:32 +00:00
henrike@webrtc.org
c2ac8953d5 Fixes Valgrind warnings in system wrappers unittest. 
Review URL: http://webrtc-codereview.appspot.com/293006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 22:46:59 +00:00
zakkhoyt@webrtc.org
a7e70b43e2 When entering fullscreen mode, the CocoaRenderView is attached as a subview to a new full screen window. 
When the class is torn down, the view was not being attached back to it's original NSView. I added a 
new class variable to remember the original superview and then reattach it at the appropriate time.
Review URL: http://webrtc-codereview.appspot.com/290009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 22:30:50 +00:00
mikhal@webrtc.org
b9db43e1b6 video_coding/jitter buffer: Reduce delay on a complete frame: No need for the next frame when current frame is already complete. 
Review URL: http://webrtc-codereview.appspot.com/289007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:38:01 +00:00
mflodman@webrtc.org
511f82eee9 Refactored ViESyncModule. 
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/291007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:31:36 +00:00
perkj@webrtc.org
68f2168978 Remove global voe::Channel::numSocketThreads. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:11:23 +00:00
mflodman@webrtc.org
27a82a65ca Refactored ViEBaseImpl. 
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/290008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:04:26 +00:00
andrew@webrtc.org
587c844741 Query the capture volume immediately on Win Core. 
This allows the capture volume to be queried immediately at capture
startup, rather than waiting the usual one second interval.

Review URL: http://webrtc-codereview.appspot.com/297003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 17:43:05 +00:00
henrik.lundin@webrtc.org
524eb48081 Removing deprecated NetEQ APIs 
Removing WebRtcNetEQ_GetPreferredBufferSize and
WebRtcNetEQ_GetCurrentDelay and all dependent APIs.

Review URL: http://webrtc-codereview.appspot.com/289006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 16:21:22 +00:00
xians@webrtc.org
0dffc6449a To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory. 
This patch disbale some codecs which are not considered necessary. 
Review URL: http://webrtc-codereview.appspot.com/299001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 15:35:44 +00:00
stefan@webrtc.org
0c2adf0b75 Fix bug introduced when enabling VP8 frame dropping. 
Also fixes two unit test mismatches.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/299002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 14:41:58 +00:00
stefan@webrtc.org
ac2c677bf6 Make all video_coding tests use the resources and output directories. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/298001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 14:23:39 +00:00
andrew@webrtc.org
d2daa5c13e Use clang by default on Mac. 
But disable Chrome clang plugins for the time being.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 01:16:06 +00:00
andrew@webrtc.org
268257475b Fix one more Objective-C clang error. 
(Analogous to r1056).

BUG=issue78

Review URL: http://webrtc-codereview.appspot.com/297004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 00:54:04 +00:00
zakkhoyt@webrtc.org
2687b261d5 Since the CocoaRenderView is forward declared with @class instead of imported, 
instance must be cast to NSView* when passed to NSView's addSubView method.
Review URL: http://webrtc-codereview.appspot.com/288001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 23:55:19 +00:00
punyabrata@webrtc.org
c9801465b6 Adding a check to ensure that the memcpy does not exceed bounds of the arrays. 
Review URL: http://webrtc-codereview.appspot.com/290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 18:49:54 +00:00
andrew@webrtc.org
1e91693fe2 Move stream_delay check to ProcessStream(). 
- was_stream_delay_set_ was being incorrectly reset in
AnalyzeReverseStream().
- Added tests to catch this case.

BUG=
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/291011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 18:28:57 +00:00
henrike@webrtc.org
0bf2ca2eed Fixes broken unit test http://code.google.com/p/webrtc/issues/detail?id=154 
Review URL: http://webrtc-codereview.appspot.com/292007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 18:21:46 +00:00
mikhal@webrtc.org
5fef05b529 libyuv: Updating paths for test files 
Review URL: http://webrtc-codereview.appspot.com/289010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 17:50:07 +00:00
mflodman@webrtc.org
ffabb59f6e Refactored ViERefCount. 
In a coming CL: Use ref count in system_wrappers instead of this class.

Review URL: http://webrtc-codereview.appspot.com/291010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 17:31:21 +00:00
henrik.lundin@webrtc.org
fc9b903fbe Enable NetEQ statistics unit testing 
Adding test target NetEqDecodingTest::TestNetworkStatistics.
Update neteq_unittest to get files from resources folder.
Update DEPS file to get resources revision 2.

Review URL: http://webrtc-codereview.appspot.com/291013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 15:38:27 +00:00
henrik.lundin@webrtc.org
2d8125dd1a Testing NetEQ network statistics 
Implementing helper function for new unit test
NetEqDecodingTest::TestNetworkStatistics. The test itself
remains to be defined. (Will be added in a coming CL.)
This change required some refactoring of the test code
to avoid excessive code duplication.

Review URL: http://webrtc-codereview.appspot.com/295009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 14:30:28 +00:00
kjellander@webrtc.org
c625c1010a Updated system_wrappers_unittests to use the test_support_main target. 
Review URL: http://webrtc-codereview.appspot.com/291012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 12:11:06 +00:00
stefan@webrtc.org
932ab18d32 Default to always NACKing residual losses when having both FEC and NACK. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/296002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 11:33:31 +00:00
bjornv@webrtc.org
4b80eb4fcd Name change resampler.c/h to aec_resampler.c/h. 
To avoid possible conflict with resampler in common_audio.
Review URL: http://webrtc-codereview.appspot.com/296006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 08:44:01 +00:00
mflodman@webrtc.org
611e4c3253 Refactored ViEPerformanceMonitor. 
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/289009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 02:39:28 +00:00
mikhal@webrtc.org
a85590d383 libyuv: Adding Android.mk 
Review URL: http://webrtc-codereview.appspot.com/291009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 01:42:57 +00:00
mflodman@webrtc.org
ad4ee3659e Refactored ViEReceiver. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 22:39:24 +00:00
marpan@webrtc.org
9d8bec6f76 FEC: Fix to valgrind warning. 
Review URL: http://webrtc-codereview.appspot.com/292009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 22:10:05 +00:00
andrew@webrtc.org
400ad6928e Fix compile warning in NS. 
BUG=issue151
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/290005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 21:33:42 +00:00
marpan@webrtc.org
d1b7932adf VP8: Setting non-zero (conservative) threshold for frame dropper. 
Review URL: http://webrtc-codereview.appspot.com/291001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 19:20:31 +00:00
mikhal@webrtc.org
2cdb2d3833 Adding Libyuv to Webrtc: 
- Adding library to DEPS file
 - Adding Wrapper implementation and tests. 

This is an interim state, as these files are not being linked at this stage.
Review URL: http://webrtc-codereview.appspot.com/259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 18:09:41 +00:00
xians@webrtc.org
e07247af8d Valgrind reports a racing condition on _sending because it is accessed by 
both TransmitMixer::PrepareDemux() and StartSend()/StopSend().
Put a lock to resolve it.
Review URL: http://webrtc-codereview.appspot.com/293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 16:31:28 +00:00
andrew@webrtc.org
1e39bc80dc Handle debug files from multiple AEC instances. 
Review URL: http://webrtc-codereview.appspot.com/295006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-27 23:46:23 +00:00
andrew@webrtc.org
a919d3a643 Don't return a zero delay with insufficient data. 
For estimating a delay over a long segment (e.g. a file) this can
throw off the estimate. Better not to add values to the AEC's histogram
until they're reliable.

BUG=
TEST=audiproc, audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/292004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-27 23:40:58 +00:00
stefan@webrtc.org
94a8c03141 Slightly increased bandwidth adaptation at both receive- and send-side. 
The send-side increase factor is increased to better follow the pace
of the receive-side estimate, while the receive-side factor is
increased to speed up adaptation.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/297002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 14:09:37 +00:00
xians@webrtc.org
8738d277a1 Valgrind detects that there are racing conditions in RTPReceiver::PacketTimeout and RTPSender 
This CL fixes two of them.
Review URL: http://webrtc-codereview.appspot.com/295005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 13:43:53 +00:00
henrik.lundin@webrtc.org
0fcc2eb368 Cleaning up neteq_unittest 
- Conforming to testing standards.
- Fixing a way of generating new reference output files.
- ifdef the test to run only on linux 64-bit
- Renaming unittest source file.
- Renaming test vectors

Review URL: http://webrtc-codereview.appspot.com/296007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 13:43:42 +00:00
henrik.lundin@webrtc.org
789da89d37 Fix a valgrind warning in NetEQ 
The special cases for packet sizes <= 10 ms (one case for each
sample rate) resulted in reading outside of the pw16_decoded
vector. This is now fixed by making sure that WebRtcSpl_DownsampleFast
gets correct input and output vector lengths.

Review URL: http://webrtc-codereview.appspot.com/295008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 12:35:31 +00:00
stefan@webrtc.org
0ee8ba1929 Remove WebRTC dependency on libvpx_lib and libvpx_include. 
Removes dependencies on libvpx_lib and libvpx_include targets when
building with Chromium.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/293004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 12:12:43 +00:00
xians@webrtc.org
83661f534e fixing the racing conditions 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 10:58:15 +00:00
henrik.lundin@webrtc.org
859626570a VP8 RTP work 
Fixing the plumbing to get the KEYIDX between VP8 wrapper and
rtp_rtcp module. Also fixing a missing pipe for temporalIdx

Review URL: http://webrtc-codereview.appspot.com/295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 10:17:00 +00:00
braveyao@webrtc.org
0a18522e1b Add support to 96kHz sampling rate to Windows CoreAudio interface. 
Review URL: http://webrtc-codereview.appspot.com/295003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 02:45:39 +00:00
mflodman@webrtc.org
26b9777e62 Only trigger one call to OnNetworkChanged for each incoming RTCP packet. 
Review URL: http://webrtc-codereview.appspot.com/289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 15:22:33 +00:00
mflodman@webrtc.org
471e83e592 Refactored ViESharedData. 
Only vie_shared_data.* are refactored, all *_impl.cc are only changed due to changed names of members in ViESharedData. These files will be refactored later, so the indentation in these files might be corrupt at this stage.

References are not changed to pointers at this stage.

Review URL: http://webrtc-codereview.appspot.com/292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 15:16:00 +00:00
henrik.lundin@webrtc.org
9af365d3c5 Fixing VP8 RTP parser bug 
Missing one initialization of new struct variable hasKeyIdx.

TBR=stefan@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/296004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 13:28:29 +00:00
henrik.lundin@webrtc.org
6f2c0168f0 Updating to VP8 RTP spec rev -02 
Updating the VP8 packetizer class (RtpFormatVp8) and VP8 parser
(in class RTPPayloadParser) to follow the -02 revision of the spec.
See http://tools.ietf.org/html/draft-ietf-payload-vp8-02.

Updating the unit tests, too. Finally, updating the tests to
follow the recommendations from the test team; specifically
including the test code in the webrtc namespace, and omitting
the main function at the end of each test file.

Review URL: http://webrtc-codereview.appspot.com/296003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 12:52:40 +00:00
mflodman@webrtc.org
6d26ef76ea Refactored ViESender. 
In a later CL:
- References -> const or ptr.

Review URL: http://webrtc-codereview.appspot.com/291003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1011 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 08:31:06 +00:00
kjellander@webrtc.org
d492f72e43 Added empty unit tests to get code coverage measured. 
In order to get code coverage recorded, there must be an executing test that is linked to the code to measure.
These projects are currently not showing up in the code coverage.

Review URL: http://webrtc-codereview.appspot.com/293002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1010 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 07:20:00 +00:00
amyfong@webrtc.org
55d81ea517 ViE Custom Call observer now using pointers, fixed protection method and miscellaneous TODO cleanup 
Review URL: http://webrtc-codereview.appspot.com/282004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1009 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 01:15:10 +00:00
andrew@webrtc.org
ba028a31c9 Fix sample rate printout in process_test. 
TBR=bjornv

Review URL: http://webrtc-codereview.appspot.com/292005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 20:37:12 +00:00
phoglund@webrtc.org
f3d10d3dfd Fixed release compilation error-warnings. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/290004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 15:56:27 +00:00
phoglund@webrtc.org
c4c56ed20b Rewrote vie_auto_test to use googletest macros. 
Removed error counting entirely - that's completely managed by googletest now, except for custom call, loopback and simulcast call.

Rewrote remaining tests to use GTest asserts.

Rewrote more tests to use GTest macros. The External Codec module is now in the build by default.

Merge branch 'master' into macro_improvements

Rewrote some more code to use GTest asserts.

The manual standard tests now also go through gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/287002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 15:23:11 +00:00
bjornv@webrtc.org
48b68c0c24 Added support for 96 kHz sampling frequency. 
Updated resampler_unittests with the new valid combinations.
Verified audio quality on files.

TEST=resampler_unittests, voe_auto_test
BUILDTYPE=Debug, Release
PLATFORM=Linux
Review URL: http://webrtc-codereview.appspot.com/294001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 13:50:41 +00:00
henrik.lundin@webrtc.org
4257790d2d NetEQ-related bug in ACM 
Fixing a bug when creating new NetEQ slave instances in ACM.
The old code called WebRtcNetEQ_GetCurrentDelay() for the
master instance to get a delay value for WebRtcNetEQ_SetExtraDelay().
This is wrong, since WebRtcNetEQ_GetCurrentDelay() reports on the
current total buffer length, while WebRtcNetEQ_SetExtraDelay() is
the extra delay that is desired to in order to sync with video.

The fix includes keeping the extra delay value in a member variable
in the ACMNetEQ class.

Review URL: http://webrtc-codereview.appspot.com/295001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 13:04:05 +00:00
kjellander@webrtc.org
543c3eaa46 Fixing Release compilation errors 
Review URL: http://webrtc-codereview.appspot.com/267026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 12:20:35 +00:00
henrik.lundin@webrtc.org
89ab652250 Cleaning up NetEQ statistics 
Removed struct MCUStats_t and all references to it.
Removed totalDiscardedPackets and totalFlushedPackets
from the PacketBuf_t struct.

Review URL: http://webrtc-codereview.appspot.com/293001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@999 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 11:06:05 +00:00
henrik.lundin@webrtc.org
df10de4b27 Removing statistics API from NetEQ 
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.

Review URL: http://webrtc-codereview.appspot.com/285002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 09:36:23 +00:00
braveyao@webrtc.org
7d3e9498bc This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases. 
Review URL: http://webrtc-codereview.appspot.com/276011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 03:35:42 +00:00
mikhal@webrtc.org
2b838b4121 video_coding: updating the session info unit test following recent changes 
Review URL: http://webrtc-codereview.appspot.com/290002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 00:20:19 +00:00
mikhal@webrtc.org
425b377973 video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update. 
Review URL: http://webrtc-codereview.appspot.com/289001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 23:41:29 +00:00
mikhal@webrtc.org
f13388f134 video_coding: Requesting a key frame after a JB flush 
Review URL: http://webrtc-codereview.appspot.com/280006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:57:51 +00:00
mikhal@webrtc.org
6b9a7f8704 video_coding: Allowing for a decodable state independent of selective nacking 
Review URL: http://webrtc-codereview.appspot.com/263001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:48:20 +00:00
andrew@webrtc.org
828af1b4b9 Add lookahead to the delay estimator. 
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/279014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:40:27 +00:00
andrew@webrtc.org
5a529395aa Make DMO init safe when not supported. 
BUG=issue133
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/284001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 18:04:26 +00:00
mflodman@webrtc.org
dfe89e337e Move ViE main/test/AutoTest to test/auto_test. 
Only paths in gyp and mk files are changed, source files are only moved.

Review URL: http://webrtc-codereview.appspot.com/267027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 14:03:15 +00:00
andrew@webrtc.org
8594f7688b Add a gyp variable for AEC debug dumps. 
TEST=process_test.cc

Review URL: http://webrtc-codereview.appspot.com/276012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 00:51:41 +00:00
kma@webrtc.org
a249f35203 Correct several makefile errors for Android build. 
Review URL: http://webrtc-codereview.appspot.com/267024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 22:16:10 +00:00
mflodman@webrtc.org
6830bdd929 Fix xcode build. 
Review URL: http://webrtc-codereview.appspot.com/280007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 15:53:15 +00:00
mflodman@webrtc.org
94ea32ef60 Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. 
Review URL: http://webrtc-codereview.appspot.com/283002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 14:49:31 +00:00
kjellander@webrtc.org
274c2efbc1 Adding empty test method required to get code coverage 
Review URL: http://webrtc-codereview.appspot.com/279008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 09:36:28 +00:00
marpan@webrtc.org
3caa327af0 VP8 wrapper: Turn on some mild amount of deblocking in post-processing. 
Review URL: http://webrtc-codereview.appspot.com/268015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-19 01:08:09 +00:00
henrike@webrtc.org
ce9d89d892 Fixes linux build error introduced in r980. 
Review URL: http://webrtc-codereview.appspot.com/279012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-19 00:14:37 +00:00
henrike@webrtc.org
ad98a3eed0 Fixes TEST crash triggered by webrtc-codereview.appspot.com/268014. 
Review URL: http://webrtc-codereview.appspot.com/280005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 23:55:12 +00:00
henrike@webrtc.org
31d30700d6 Addressed review comments from http://webrtc-codereview.appspot.com/256004/ 
Review URL: http://webrtc-codereview.appspot.com/256007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 19:59:32 +00:00
kma@webrtc.org
ced118636d Changed keyword __restrict__ to __restrict. 
Review URL: http://webrtc-codereview.appspot.com/279011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 17:51:19 +00:00