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589 Commits

Author SHA1 Message Date
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
stefan@webrtc.org
7de07652ad Disables a flaky metric test. 
This is a duplication of issue 255008 since I wasn't able to commit that one
from the computer on which it was created.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/276007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-14 15:16:16 +00:00
tommi@webrtc.org
ded85f14ef Enable WEBRTC_NO_TRACE for Chromium builds. 
I'm also fixing WEBRTC_TRACE so that it won't break the build but on Linux I had to do something non traditional as is explained in the comments.
Review URL: http://webrtc-codereview.appspot.com/269012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-14 09:39:31 +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
stefan@webrtc.org
b351d6a8d8 Reverting rev 929 due to failing assert on Linux. 
Failing at: audio_buffer.cc:159

TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/270008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 13:26:05 +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
niklas.enbom@webrtc.org
50b3cbe979 First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable. 
Review URL: http://webrtc-codereview.appspot.com/269007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 08:31:32 +00:00
kma@webrtc.org
b61c410347 Fixed a couple of Android makefiles to let voe and vie build properly. 
Review URL: http://webrtc-codereview.appspot.com/278001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 18:10:25 +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
d30b688751 Remove TraceScan executable. 
Review URL: http://webrtc-codereview.appspot.com/270002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 22:23:20 +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
niklas.enbom@webrtc.org
f3c1b87f00 my eyes started bleeding when I saw this... 
Review URL: http://webrtc-codereview.appspot.com/268005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 12:43:48 +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
niklas.enbom@webrtc.org
4cd841e9a6 Fix win compile error for interpolator_test 
Review URL: http://webrtc-codereview.appspot.com/269003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 08:02:16 +00:00
phoglund@webrtc.org
cff98ca6ff Made it possible to run the voe_auto_test standard test in GTest behind a flag. The purpose is to run the whole test without any manual intervention since we want to run the test on a build bot in automated mode. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/267001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-08 13:08:25 +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
andrew@webrtc.org
b353d21560 ...and now fix the Debug build. 
Review URL: http://webrtc-codereview.appspot.com/272001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-05 00:57:33 +00:00
andrew@webrtc.org
369766ed29 Fix Release mode errors in common_video tests. 
Review URL: http://webrtc-codereview.appspot.com/271001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 23:59:56 +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
andrew@webrtc.org
a4b9660372 Add mistakenly removed VAD enabling function. 
This resolves the unknown VAD status warnings introduced in r845.

BUG=
TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 01:36:27 +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