generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
1,628 Commits 1 Branch 0 Tags 108 MiB
145f04f0c4
Commit Graph

816 Commits

Author SHA1 Message Date
stefan@webrtc.org
efd0a48c61 Add error resilient mode options to the VP8 specific VideoCodec struct. 
It is useful to disable error resilience when we know we won't decode
with errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-29 10:12:35 +00:00
andrew@webrtc.org
6d6a43d6e3 Use char as ring-buffer data type. 
- Avoids a bunch of char* casts.
- Use enum type rather than char.

TEST=audioproc_unittest on Linux (float and fixed), build on Windows

Review URL: http://webrtc-codereview.appspot.com/336010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-28 22:40:15 +00:00
bjornv@webrtc.org
267d0133ff Fixed pointer operations on void. 
This should fix the error on Win where pointer arithmetics are done on void pointers. Type cast to char to interpret a size.
Review URL: http://webrtc-codereview.appspot.com/329019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-28 10:26:17 +00:00
bjornv@webrtc.org
7270a6bcc2 Merged apm-buffer branch [r1293] back to trunk. 
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.

TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-28 08:44:17 +00:00
mikhal@webrtc.org
e39de16fa5 Moving video type convert functionality to libyuv. deleting vplibConversions as it is no longer needed. 
Review URL: http://webrtc-codereview.appspot.com/338002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-27 23:45:30 +00:00
stefan@webrtc.org
f6c6b1c5b5 Include the media packet FEC headers in the video bitrate. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-23 10:33:39 +00:00
stefan@webrtc.org
39670f6aa6 Only reset the last decoded sequence number after flushing until key frame. 
We can't reset the complete last decoded state when we recycle until a
key frame because that will allow any delta frame to be decoded afterwards,
and since the decoder isn't reset we will get decode errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/330003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-23 09:08:51 +00:00
mflodman@webrtc.org
1ce66e4dfb Don't report error when failing to send RTCP BYE. 
Review URL: http://webrtc-codereview.appspot.com/337002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 18:40:15 +00:00
stefan@webrtc.org
6a4bef4e65 Implements selective retransmissions. 
Default is set to not retransmit VP8 non-base layer packets or FEC packets.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:52:41 +00:00
pwestin@webrtc.org
f4d3b9d5a1 Cleaned up leaky symbols in NS. 
Review URL: http://webrtc-codereview.appspot.com/337001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:33:08 +00:00
pwestin@webrtc.org
ebcb6421b1 Cleaned up leaky symbols in G722. 
Review URL: http://webrtc-codereview.appspot.com/333017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:20:06 +00:00
pwestin@webrtc.org
d8f8b32521 Cleaned up leaky symbols in iSAC. 
Review URL: http://webrtc-codereview.appspot.com/329014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:19:43 +00:00
mflodman@webrtc.org
84dc3d134d Add REMB functionality to ViE. 
This CL only adds support for encoding one stream, but receiving multiple streams.

BUG=
TEST=video_engine_core_unittest + auto_test/loopback

Review URL: http://webrtc-codereview.appspot.com/333011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 10:26:13 +00:00
pwestin@webrtc.org
093ffad26b Removed unused function messing up the symbols. 
Review URL: http://webrtc-codereview.appspot.com/336006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 09:48:48 +00:00
henrik.lundin@webrtc.org
1e28d3c2e1 Change VP8 packetizer to use a single max payload size 
The packetizer class is changed so that the max payload size is
provided on construction of the class rather than for each packet.
The tests are re-written to comply with the new design.

Also fixing a few errors in the tests.

Review URL: http://webrtc-codereview.appspot.com/335010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:49:31 +00:00
stefan@webrtc.org
f5edb923b1 Remove unused variable. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:34:31 +00:00
pwestin@webrtc.org
8edb39db30 Prevent sending empty RTCP packet. 
Review URL: http://webrtc-codereview.appspot.com/331009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 07:40:33 +00:00
henrik.lundin@webrtc.org
4a19030131 New VCM robustness API 
This CL defines and starts to implement a new robustness API for
video coding module. The API is partly implemented. Some of the
modes and methods are still TBD.

Also including a new unittest with mocking of decoder and callbacks,
and faking of system clock.

Review URL: http://webrtc-codereview.appspot.com/333006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 20:38:37 +00:00
andrew@webrtc.org
697bc43b67 Restore item deletions in Windows UDP. 
TEST=voe_auto_test on Windows.

Review URL: http://webrtc-codereview.appspot.com/331013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 19:58:11 +00:00
andrew@webrtc.org
71571c5446 Remove unneeded variables from windows UDP. 
TEST=build on Windows.

Review URL: http://webrtc-codereview.appspot.com/329013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 18:30:59 +00:00
mallinath@webrtc.org
03532b5f41 Fixing the double delete problem in UdpSocket2ManagerWindow. PopFront deletes the items, to there is no need to delete item explicitly. 
Review URL: http://webrtc-codereview.appspot.com/333014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 15:36:44 +00:00
henrik.lundin@webrtc.org
7d8c72e2db Re-implement dependency injection of TickTime into VCM and tests 
This change basicly re-enables the change of r1220, which was
reverted in r1235 due to Clang issues.

The difference from r1220 is that the TickTimeInterface was
renamed to TickTimeClass, and no longer inherits from TickTime.

Review URL: http://webrtc-codereview.appspot.com/335006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 15:24:01 +00:00
kjellander@webrtc.org
5490c71a1b Converted to gtest, writing output files properly and no longer uses exceptions. 
This test now runs and fails as a gtest should (previously it always
exited with 0 even if the tests failed).
The audio_coding_module_test target no longer uses exceptions in the generated project.
Output files are written to our global output folder, using
testsupport/fileutils.h.

BUG=
TEST=audio_coding_module_test on all platforms, in Debug+Release

Review URL: http://webrtc-codereview.appspot.com/334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 13:34:18 +00:00
stefan@webrtc.org
898f881e32 Make sure the next frame to be decoded is cleaned up if it's empty. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/332001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 09:13:14 +00:00
niklas.enbom@webrtc.org
6c9be123ef Letting strncpy do its job. Landing and extending http://webrtc-codereview.appspot.com/329010/ on behalf of tbreisacher. 
Review URL: http://webrtc-codereview.appspot.com/335009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:59:31 +00:00
stefan@webrtc.org
8c5d24266e Fix VP8 layer 2 sync dependencies. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:56:04 +00:00
henrik.lundin@webrtc.org
00e730730e Refactoring RtpFormatVp8Test 
This is the first change in a series of changes to get new functionality
into the VP8 packetizer.

This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.

Review URL: http://webrtc-codereview.appspot.com/304009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:51:36 +00:00
mikhal@webrtc.org
61045a4a03 video_coding/jitter_buffer: Account for layer info when searching for the next frame 
Review URL: http://webrtc-codereview.appspot.com/328003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 23:24:19 +00:00
andrew@webrtc.org
a38ce09919 Fix last Mac/clang compile error. 
Fixes "receiver is a forward class and corresponding @interface may
not exist" error.

TEST=build on Mac with -Werror enabled.
TBR=zakkhoyt@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/333012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 22:23:21 +00:00
pwestin@webrtc.org
061fa5b828 Changed handling of padding data. 
Review URL: http://webrtc-codereview.appspot.com/331008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:56:17 +00:00
henrik.lundin@webrtc.org
dbba1f969f Packet waiting-time statistics 
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.

Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.

Updating common_types.h and VoiceEngine tests to include the
new metrics.

Unit tests are also added for NetEQ and AcmNetEq.

Review URL: http://webrtc-codereview.appspot.com/328011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:45:05 +00:00
henrik.lundin@webrtc.org
219acc6cec Including Brighten function in namespace VideoProcessing 
This change is in response to Issue 173.

BUG=http://code.google.com/p/webrtc/issues/detail?id=173

Review URL: http://webrtc-codereview.appspot.com/328012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:33:49 +00:00
stefan@webrtc.org
62fdc42e9c Fix build issue with clang. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/330009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:27:09 +00:00
stefan@webrtc.org
8dc9e4760e Fixes for selective NACKing. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/332007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:12:50 +00:00
tina.legrand@webrtc.org
5efcad1758 We used the wrong syntax for "new", which generated a warning/error building with clang. 
Review URL: http://webrtc-codereview.appspot.com/336003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:05:55 +00:00
mikhal@webrtc.org
0e7d9d862a Adding layer info consideration when applying FEC protection. In this first version, we hard code protection zero for non-base layer frames. As a future enhancement, an array should be passed from mediaOpt to set the protection per layer. A TODO was added in MediaOpt. 
Review URL: http://webrtc-codereview.appspot.com/330005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 19:04:49 +00:00
mikhal@webrtc.org
190e88a6d3 video_coding: When in hybrid mode, don't NACK non-base layer packets 
Review URL: http://webrtc-codereview.appspot.com/334002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 18:57:14 +00:00
mikhal@webrtc.org
884d8e7f4b video_coding: Updating sync state based on the layer flag 
Review URL: http://webrtc-codereview.appspot.com/333004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 18:53:05 +00:00
henrik.lundin@webrtc.org
303158588b Revert "Inject TickTimeInterface into VCM and tests" 
This CL reverts r1220.

Review URL: http://webrtc-codereview.appspot.com/336002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:55:45 +00:00
henrika@webrtc.org
e32c08a5a6 Removes usage of default parameters and fixes a bug which was found 
using Clang on Linux.

BUG=none
TEST=none
TBR=pwestin

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:39:48 +00:00
stefan@webrtc.org
b33f9dccd6 Correction to how the VP8 wrapper generates picture ids. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 10:24:02 +00:00
tina.legrand@webrtc.org
398af2337b Solving issue 178, errorbuild warnings on Mac. 
This CL continues the work of solving issue 178. A small change in one file.
Review URL: http://webrtc-codereview.appspot.com/330006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 07:36:07 +00:00
henrike@webrtc.org
cf5bcd1fd2 Removed usage of the deprecated critical section constructor in audio_conference_mixer. 
Review URL: http://webrtc-codereview.appspot.com/320007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 23:00:30 +00:00
andrew@webrtc.org
8a44259ea8 Move static consts out of class. 
Still causing a gtest error on non-Win platforms. This should fix it...

TBR=asapersson@webrtc.org
TEST=build on Linux

Review URL: http://webrtc-codereview.appspot.com/332006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 21:24:30 +00:00
andrew@webrtc.org
41192469f6 Switch enums to consts to fix gtest error. 
TBR=asapersson@webrtc.org
TEST=build on Windows

Review URL: http://webrtc-codereview.appspot.com/330008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1224 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 20:55:46 +00:00
henrike@webrtc.org
105e07193e Removed usage of the deprecated critical section constructor in modules/utility. 
Review URL: http://webrtc-codereview.appspot.com/321006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 19:53:46 +00:00
marpan@webrtc.org
57353a33f1 FEC Receiver: Fix to how old packets (e.g., re-tranmitted packets in hybird NACK-FEC mode) are treated. 
This change avoids having old packets end up on the current packet list for FEC decoding, and so they are immediately sent out to jitter buffer.
The current list of packets for FEC decoding are sent out only when new packet arrives (with time-stamp greater than current).
Review URL: http://webrtc-codereview.appspot.com/322009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 17:21:09 +00:00
henrik.lundin@webrtc.org
e7d8c56c56 Fix for dual decoder in VCM receiver 
In VCMReceiver::FrameForDecoding, one of the if-cases could sometimes
extract an incomplete frame without first copying the state to the
dual decoder.

Review URL: http://webrtc-codereview.appspot.com/328006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 15:40:52 +00:00
henrik.lundin@webrtc.org
a70f945086 Inject TickTimeInterface into VCM and tests 
The purpose of this change is to introduce dependency injection
of the timer into the video coding module.

Review URL: http://webrtc-codereview.appspot.com/332003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 14:40:05 +00:00
asapersson@webrtc.org
5249cc8f77 Review URL: http://webrtc-codereview.appspot.com/295010 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 14:31:37 +00:00
tina.legrand@webrtc.org
9775a30859 Added variable to catch return value. 
Review URL: http://webrtc-codereview.appspot.com/329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 11:15:46 +00:00
kjellander@webrtc.org
08dec7f449 Now using fileutils.h OutputPath to write output to the right directory and ResourcePath to read resource files from the resource bundle. 
Removed some Valgrind warnings by closing output files. There are still some Valgrind warnings left, that needs to be fixed by a developer with more insight.

Updated all include paths to contain full paths to header files.

Tested in Debug+Release on Linux, Mac and Windows.
All tests ran successfully except the VideoProcessingModuleTest.ContentAnalysis test that fails on Windows with the following error:
unknown file: error: SEH exception with code 0xc0000005
thrown in the test body.
Fixing that is out of scope for this CL.

Review URL: http://webrtc-codereview.appspot.com/266011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 10:31:38 +00:00
tina.legrand@webrtc.org
554ae1ad4e Changes to solve warnings on Mac, issue #178. 
Review URL: http://webrtc-codereview.appspot.com/320005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 10:09:04 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
kjellander@webrtc.org
543611a77a Reverting r972 due to compilation error on Windows Release build. 
TBR=kma
Review URL: http://webrtc-codereview.appspot.com/282003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 13:25:13 +00:00
bjornv@webrtc.org
2f047ccede Removed unnecessary variable to avoid compiler error on Win. 
Review URL: http://webrtc-codereview.appspot.com/267021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 12:03:25 +00:00
henrik.lundin@webrtc.org
ba74924043 Remove use of exceptions in NetEQ test code 
Replaced the exceptions thrown when codec instance creation failed
with simple exit(EXIT_FAILURE). There is no point in continuing
if creating the codec fails.

Review URL: http://webrtc-codereview.appspot.com/282002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 09:55:01 +00:00
bjornv@webrtc.org
6a9835d59c Delay estimator structural changes. 
Improved the way we handle different data types (float vs fixed) and reduced the complexity by nearly 50%.
We now have a generic struct for both float and fixed delay estimators and a core struct for the binary spectrum based delay estimator. All wrapper codes (for both fixed and float) are gathered in delay_estimator_wrappers.*.
Moved out the far end history buffer to AEC(M).
Added a union to handle difference types when create.
Review URL: http://webrtc-codereview.appspot.com/277004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 08:30:34 +00:00
kma@webrtc.org
fa9b016fb5 Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix. 
(1) For generic platforms, code was changed to remove the shifting within loops.
Basically, it's just change a loop from
  for() {
    sum += (a*b) >> scale;
  }
to:
  for() {
    sum += (a*b);
  }
  sum >> scale;

Type int64_t is used for sum to make sure no information is not lost.
Performance is about the same as before the change. Bits are not exact,
although in theory the change should have preserved more information. The purpose
of this change is to make the generic code and ARM code bit exact, simpify the code,
while keep the speech quality at least not lower. (Some speech tests might be good.)

(2) For ARM platform, used assembly to optimize the performance. iSAC runs faster
with this change. (Reduced run time of an offline file test from 10.16ms to 8.81ms)
Review URL: http://webrtc-codereview.appspot.com/267014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 02:50:55 +00:00
braveyao@webrtc.org
f556b9d1f4 This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend(). 
Review URL: http://webrtc-codereview.appspot.com/277007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 02:17:28 +00:00
kjellander@webrtc.org
cd7b57ef9e Fixing release compilation error 
Review URL: http://webrtc-codereview.appspot.com/279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 14:26:21 +00:00
kjellander@webrtc.org
3f1cb8e546 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/269018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:56:54 +00:00
kjellander@webrtc.org
cc2ecb3c2e Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/267019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:48:36 +00:00
kjellander@webrtc.org
b72268e147 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/280004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:39:15 +00:00
kjellander@webrtc.org
64a897a772 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/282001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:33:11 +00:00
kjellander@webrtc.org
c05b56a38b Fixing compilation error 
Review URL: http://webrtc-codereview.appspot.com/276010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 08:59:48 +00:00
kjellander@webrtc.org
0403ef419f Restructuring and adding unit test targets on project level instead of in common_audio. 
Review URL: http://webrtc-codereview.appspot.com/280001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 08:35:47 +00:00
phoglund@webrtc.org
337dc68992 Included modules in webrtc.gyp and fixed build errors. 
Removed TODO from webrtc.gyp since it is done.

Tabs -> spaces.

Tabs -> spaces.

Tabs -> spaces.

Fixed compilation on Windows.

Added missing file.

Merge branch 'master' into fix_mac_modules

Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings.

 Please enter the commit message for your changes. Lines starting

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 15:36:44 +00:00
stefan@webrtc.org
fcf33eb7e0 Limit number of send-side BWE increases to one per second. 
Also report 0 losses if not enough expected packets since
previous receiver report.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/270009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 07:58:31 +00:00
punyabrata@webrtc.org
81d4499dee Microphone volume on Mac not being printed properly due 
to a mismatch in variable type. Additionally, now printing
a volume that will range from 0 - 255
Review URL: http://webrtc-codereview.appspot.com/267016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 02:06:49 +00:00
andrew@webrtc.org
755b04a06e Add RMS computation for the RTP level indicator. 
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
  AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.

TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/279003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 16:57:56 +00:00
andrew@webrtc.org
6a85b17a0a Potential fix for crash after Mac sleep. 
When a Mac goes to sleep, the OS pauses the IO threads. If a
subsequent StopSend/Playout happens, we time out waiting for the IO
threads, but didn't ensure they were shut down.

BUG=
TEST=voe_cmd_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/269013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 16:23:41 +00:00
kjellander@webrtc.org
85596d5bf4 Setting completeFrame to true for all created encoded images. 
Review URL: http://webrtc-codereview.appspot.com/276008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 13:45:25 +00:00
henrik.lundin@webrtc.org
bc91d5af86 NetEQ tests 
Adding capability to parse RED payloads to the RTPanalyze tool.
Also adding a method to scramble an RTP payload (currently not
used).

Review URL: http://webrtc-codereview.appspot.com/276006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 10:16:01 +00:00
mflodman@webrtc.org
a02ef1ace2 Fix broken tree. 
Review URL: http://webrtc-codereview.appspot.com/267015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 07:50:50 +00:00
mflodman@webrtc.org
1f69c03739 Added size sanity check for copying app specific RTCP data. 
Similar check as done in RTCPUtility::RTCPParserV2::ParseAPPItem.

Review URL: http://webrtc-codereview.appspot.com/277002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 06:12:39 +00:00
henrik.lundin@webrtc.org
33df5335bf Change luminance of all pixels by a specified value. 
Modeled on color_enhancement.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269004
Patch from SriRam <tvnsriram@google.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-14 15:30:26 +00:00
andrew@webrtc.org
0db7dc6e18 Add file-playing channels to voe_cmd_test. 
Fix file reading and writing.

TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/279001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-13 01:34:05 +00:00
andrew@webrtc.org
cd8243807e Unpack the full set of audioproc data. 
Review URL: http://webrtc-codereview.appspot.com/276004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 19:13:36 +00:00
kma@webrtc.org
d71d480487 Fixed a build error of audio conference mixer in Android. 
Review URL: http://webrtc-codereview.appspot.com/267009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 17:14:23 +00:00
mflodman@webrtc.org
fd3a0efd15 RTP bw estimate fix. 
Review URL: http://webrtc-codereview.appspot.com/279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 10:55:26 +00:00
phoglund@webrtc.org
1144ba2268 Base and codec tests now run verify output and render to file instead of to screen. 
Rewrote the codec test to render to file and do video comparisons.

Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.

Added video analysis to the test. This will make sure that the system output roughly the right thing.

Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.

Made sure no one passes in too large YUV videos into the autotest.

The standard test's output now gets captured for both the left and right windows.

Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/249001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 09:01:03 +00:00
kma@webrtc.org
13318ef422 (1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing. 
(2) Added a makefile for testing fiexed point iSAC in Android.
Review URL: http://webrtc-codereview.appspot.com/266005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 18:00:22 +00:00
mflodman@webrtc.org
7a4eb2837a Calculate the available bandwidth before sending a TMMBR 
Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.

Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:54:46 +00:00
mflodman@webrtc.org
637a59e68e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined. 
Review URL: http://webrtc-codereview.appspot.com/266010
Patch from mikhals <mikhal@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:50:48 +00:00
tina.legrand@webrtc.org
855a77c972 Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo. 
Solving issue 130 reported by Niklas.

Reviewer: Turaj
Review URL: http://webrtc-codereview.appspot.com/268007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 08:17:08 +00:00
andrew@webrtc.org
c4f129f97c Improve the mixing saturation protection scheme. 
A single participant is not processed at all. With multiple
participants, we divide-by-2 as before when mixing. Afterwards,
the mixed signal is limited by the AGC to -7 dBFS and then doubled to
restore the original level.

This preserves the level while guaranteeing good saturation protection.

Add a test to voe_auto_test. Hijack and improve the existing mixing test
for this.

TEST=voe_auto_test, voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/241013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 03:41:22 +00:00
andrew@webrtc.org
4b13fc9c09 Add delay modification to process_test. 
Review URL: http://webrtc-codereview.appspot.com/266007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 19:27:11 +00:00
henrike@webrtc.org
2f32b5c8a7 Fixes an issue where file playing could happen at a lower sampling frequency than the file. 
Details:
The mixer looks at all the participants desired frequency and concludes the highest desired mixing frequency. This is the frequency that the mixer will mix at. Participants that are always mixed are in a separate list and the function concluding the highest desired mixing frequency did not look at that list and therefore always conclude that the lowest mixing frequency is sufficient.
Review URL: http://webrtc-codereview.appspot.com/277003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 19:02:17 +00:00
mikhal@webrtc.org
eb4ef17bbd Removing vplib include and VideoInterpolator when not needed 
Review URL: http://webrtc-codereview.appspot.com/268004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 18:11:02 +00:00
kjellander@webrtc.org
488ed92c3b Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/267003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:12:40 +00:00
kjellander@webrtc.org
c3a4dcd101 Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/266008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:11:44 +00:00
kjellander@webrtc.org
ad79d6f164 Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/267002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:11:14 +00:00
mflodman@webrtc.org
03a9eb1526 RTP module: Make sure payloadName is null terminated. 
Review URL: http://webrtc-codereview.appspot.com/268006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 14:51:18 +00:00
kjellander@webrtc.org
9dcab8fb14 Restoring Android.mk 
This is the last file left from 256006 that I forgot to restore according to your comments.
The other Android.mk you fixed in 266004.

Review URL: http://webrtc-codereview.appspot.com/268003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 08:59:13 +00:00
henrikg@webrtc.org
c58ef08da2 Removes system CPU measurement for Chrome build. 
It does not work on Chrome Windows, and is anyway not needed for Chrome.
Review URL: http://webrtc-codereview.appspot.com/243006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-08 08:44:17 +00:00
henrik.lundin@webrtc.org
f15fbc379d Change in RTP module SendVP8 
Changing how the max payload length is calculated. Instead
of handling RTP and FEC header overhead explicitly, call the
MaxDataPayloadLength method which already does it. Avoid redundant code. Had to move MaxDataPayloadLength to the
RTPSenderInterface.

Review URL: http://webrtc-codereview.appspot.com/269002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-08 08:23:47 +00:00
kma@webrtc.org
9b813510eb Changes for building audio coding in anroid. Only makefiles are touched. 
Review URL: http://webrtc-codereview.appspot.com/266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 23:30:01 +00:00
henrike@webrtc.org
26d3667a26 Fix for broken test after r897 
Review URL: http://webrtc-codereview.appspot.com/274001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 23:24:40 +00:00
henrike@webrtc.org
e2a34f8275 Removes the API for setting RX VAD since the RX vad should always be on anyways. 
Review URL: http://webrtc-codereview.appspot.com/264001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 21:33:24 +00:00
mflodman@webrtc.org
5ae9f5ed6c Adding logs in RTPSender::ReSendToNetwork. 
Review URL: http://webrtc-codereview.appspot.com/273001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 20:03:00 +00:00
kjellander@webrtc.org
bf483844af Restructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files. 
I also fixed compilation on Mac (by enabling exceptions for the NetEqTestTools target). Executing the test fails on Mac, but I assume this is because it checks bit exactness, similar to the issue we had with audio_coding_module (see issue 114)

Review URL: http://webrtc-codereview.appspot.com/255004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 16:05:19 +00:00
kjellander@webrtc.org
36e1ad9b5d Restructuring and removing ilbc_test.gypi. 
According to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.

No changes at all are being made in the source files; they are just moved.
The only modified files are the GYP file and Android.mk

Kevin: I updated relative paths in Android.mk so please verify it is correct, since I don't know how to build that.

Review URL: http://webrtc-codereview.appspot.com/256006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 15:27:11 +00:00
vikasmarwaha@webrtc.org
a5c4c1f1d4 Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor. 
Review URL: http://webrtc-codereview.appspot.com/253008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 23:22:51 +00:00
marpan@webrtc.org
040cb71e0a Fix windows compilation errors and warning for test_fec. Disabled VERBOSE_OUTPUT. 
Review URL: http://webrtc-codereview.appspot.com/253005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 22:57:56 +00:00
tina.legrand@webrtc.org
731e9aea79 Fixes ACM API test to build on 32-bits machines. 
Changing counters from unsigned int64 to int.
Review URL: http://webrtc-codereview.appspot.com/256010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 07:34:22 +00:00
kjellander@webrtc.org
20a370e875 Changing the namespace of TestSuite to webrtc::test. 
Adding gmock initialization into main test runner class

Review URL: http://webrtc-codereview.appspot.com/254004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 01:19:16 +00:00
kjellander@webrtc.org
1a8d08ad76 Changing usage of gtest_main target, to use test_support_main instead. 
Review URL: http://webrtc-codereview.appspot.com/252002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 23:28:47 +00:00
andrew@webrtc.org
89088b963e Fix the path to protoc.gypi. 
It was mistakenly pointing to the trunk/build rather than the
trunk/src/build copy, causing the Chrome build to fail.

TEST=./build/gyp_chromium in Chrome

Review URL: http://webrtc-codereview.appspot.com/253006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 20:43:45 +00:00
tina.legrand@webrtc.org
2475a1953a Committing a file that was part of CL 175002, but for wome reason weren't uploaded correctly. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 17:54:27 +00:00
tina.legrand@webrtc.org
fb389e3b02 This CL is divided in several patches, to make review easier. 
Patch Set 1: Removing blanks at end of lines.
Patch Set 2: Removing tabs.
Patch Set 3: Fixing include-guards.
Patch Set 4-7: Formatting files in the list.
Patch Set 8: Formatting CNG.

Patch Set 9: 
* Fixing comments from code review
* Fixing formating in acm_dtmf_playout.cc
* Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing.
* Refactored constructor of ACMGenericCodec. Rest of file still to be fixed.
* Fixing break; after return ...; in several files.

Patch Set 10:
* Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc
NOTE! Not all files have the right format. That work will continue in separate CLs.

Review URL: http://webrtc-codereview.appspot.com/175002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 17:20:10 +00:00
mikhal@webrtc.org
e203de7ba2 jitter_buffer updates: 
1. Determining continuity based on pictureId and not seq. numbers when available.
2. Hybrid bug fix: Don't set to decodable when the nack list is empty.
Review URL: http://webrtc-codereview.appspot.com/255001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:42:52 +00:00
pwestin@webrtc.org
7232ad78b2 reverted back the sanity and changed the test 
Review URL: http://webrtc-codereview.appspot.com/254006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:36:32 +00:00
pwestin@webrtc.org
cfc1070586 Fixed sanity for min length 
Review URL: http://webrtc-codereview.appspot.com/259003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:15:44 +00:00
pwestin@webrtc.org
075e91fa27 Added parsing of width and height from VP8 header 
Review URL: http://webrtc-codereview.appspot.com/241012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 23:14:58 +00:00
henrik.lundin@webrtc.org
679cb07980 Fix build error for release build 
Review URL: http://webrtc-codereview.appspot.com/252003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 19:52:27 +00:00
henrik.lundin@webrtc.org
baf6db5ead Making dual decoder work again in VCM 
Changing the assignment operator in VCMJitterBuffer to a named
function (CopyFrom) instead, since it is not a straight
assignment. Also fixing two bugs in the jitter copy function.

Bug fix in VCMEncodedFrame: The copy constructor did not
copy _length.

In VCM codec database, make sure that the callback object is
preserved when copying back from secondary to primary decoder.

In VP8 wrapper, adding code to copy the _decodedImage to the
Copy() method.

Bugfix in video_coding_test rtp_player:
The retransmissions where made in reverse order. Now new items are
appended to the end of the LostPackets list, which makes the order
correct when retransmitting.

Handling the case when cloning an unused decoder state:
When the decoder has not successfully decoded a frame yet,
it cannot be cloned. A NULL pointer will be returned all
the way out to VideoCodingModuleImpl::Decode(). When this
happens, the VCM will call Reset() for the dual receiver,
in order to reset the state to kPassive.

Review URL: http://webrtc-codereview.appspot.com/239010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 18:58:39 +00:00
kjellander@webrtc.org
d292b9c9da Unit tests now compile and run at all platforms. 
Cosmetic changes to mocks.h.

Review URL: http://webrtc-codereview.appspot.com/253003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 16:34:52 +00:00
henrik.lundin@webrtc.org
895870b68f Adding marker bit to RTPanalyze results 
Review URL: http://webrtc-codereview.appspot.com/254005

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:29:34 +00:00