Commit Graph

1150 Commits

Author SHA1 Message Date
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
mallinath@webrtc.org
2a61e15bff PortAllocator is now passed to PeerConnection instead of PeerConnectionFactory in new libjingle release.
Also creator of PortAllocator is responsible for deletion instead of factory.
DEPS file has new libjingle rivison.
Review URL: http://webrtc-codereview.appspot.com/317006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1172 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 19:12:46 +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
andrew@webrtc.org
981d74ddf4 Roll libyuv to r101.
- Fixes a build error on Linux ia32.
- Fixes a gyp warning on Windows.

Review URL: http://webrtc-codereview.appspot.com/318007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1158 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 05:48:56 +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
6073de6632 Fix libjingle Win source paths in peerconnection.
TBR=tommi@webrtc.org
TEST=Windows build

Review URL: http://webrtc-codereview.appspot.com/322005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1153 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 22:52:21 +00:00
andrew@webrtc.org
5410ef866d Rolling back to libyuv r98.
r99 breaks the 32-bit build.

TBR=mikhal@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/321007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1152 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 22:25:14 +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
kjellander@webrtc.org
480352ecf7 Fixing paths to Video engine Android.mk files.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1149 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 15:45:23 +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