Commit Graph

695 Commits

Author SHA1 Message Date
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
tommi@webrtc.org
cde1e7f42a Use a TraceNoop instance when tracing disabled (to be used in Chromium).
I'm also adding an empty implementation for static methods in the Trace
interface since the default implementation relies on TraceImpl.
Review URL: http://webrtc-codereview.appspot.com/267013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@946 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 12:23:36 +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
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
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
kma@webrtc.org
4bb141078f A change to Android makefile for building voe auto test.
Review URL: http://webrtc-codereview.appspot.com/255007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@872 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 16:38:22 +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
niklas.enbom@webrtc.org
0ba31331a8 Aligning license file with file header
git-svn-id: http://webrtc.googlecode.com/svn/trunk@868 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 09:31:39 +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
mflodman@webrtc.org
29d75b3f7d Only allow increasing capture time.
Review URL: http://webrtc-codereview.appspot.com/259001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@861 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:10:49 +00:00
andrew@webrtc.org
18ee6ec8e9 Use __inline in NS-fixed.
The use of "inline" was failing to build on Windows.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/255003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@860 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:07:46 +00:00
andrew@webrtc.org
3119ecfec8 Fix audioproc build errors on Windows.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/254003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@859 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:00:18 +00:00
mikhal@webrtc.org
c4ab8706f4 video_processing: Adding logic to avoid a memcpy when not required
Review URL: http://webrtc-codereview.appspot.com/255002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@858 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 16:44:24 +00:00
punyabrata@webrtc.org
0ab521f754 Resolving a crash related to strncopy followed by a strcat
call. strncopy will not explicity copy or add a "\0" therefore
strcat did not know where to append the "\n" which was causing
an out of bounds crash.
Because we are checking the length, strcpy should be good enough
as it also copies the "\0". Please note that that I am pre-emptively
adding 2 instead of 1 to the length to take into account of the \n
that will be added later.
Review URL: http://webrtc-codereview.appspot.com/253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@857 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 15:19:44 +00:00
kjellander@webrtc.org
d6837709cf Fixing VPMUnitTest compilation error on Windows.
It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.

Review URL: http://webrtc-codereview.appspot.com/257002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@854 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 01:51:10 +00:00
henrike@webrtc.org
b37c628ae4 Fixes crash due to r841.
Review URL: http://webrtc-codereview.appspot.com/256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 23:53:04 +00:00
kma@webrtc.org
e9f909b575 Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it.
Review URL: http://webrtc-codereview.appspot.com/244002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@852 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 22:24:57 +00:00
andrew@webrtc.org
f1a45d77fb Add missing <stdlib.h> to data_log test.
BUG=
TEST=system_wrappers_unittests

Review URL: http://webrtc-codereview.appspot.com/256002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@851 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:44:54 +00:00
andrew@webrtc.org
3134aacd6b Use fileutils for the audio file in voe_auto_test.
BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/250010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@850 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:31:07 +00:00
kma@webrtc.org
27957508a3 Changed Android makefile to make the lastest video render code run.
Review URL: http://webrtc-codereview.appspot.com/247005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@849 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:29:50 +00:00
kjellander@webrtc.org
84736882ad Fixing system_wrappers unittests.
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@848 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 20:44:24 +00:00
andrew@webrtc.org
2c74bab8b9 Remove unneeded assert and tracing.
This is related to r840.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/239019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@845 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 19:54:20 +00:00
amyfong@webrtc.org
299e2c9ea4 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame
- this caused vie_auto_test to fail for Windows (but fine for Linux & Mac).
Review URL: http://webrtc-codereview.appspot.com/253001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@844 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 19:10:26 +00:00
kjellander@webrtc.org
177bb523bd Fixing system_wrappers unittests.
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@842 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 17:10:01 +00:00
henrike@webrtc.org
066f9e5a2f Ray, please verify that this cl fixes the issue. Once the verification has been made, please review:
Henrik A: VoE
Andrew: audio_conference_mixer

Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 23:15:47 +00:00
henrike@webrtc.org
731ecba47d Review URL: http://webrtc-codereview.appspot.com/251002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@840 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 22:49:24 +00:00
braveyao@webrtc.org
1f6d740571 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout().
It could happen that if you want to restart playout, the new sponsored Render thread would catch this event
if the previous Render thread quits before this event is set.
With this modification, the device plugging out/in during talking would be supported well.
Review URL: http://webrtc-codereview.appspot.com/248002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@839 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 21:30:30 +00:00
wu@webrtc.org
88e0a34815 Remove duplicated code.
Review URL: http://webrtc-codereview.appspot.com/251001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@838 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 17:29:44 +00:00
stefan@webrtc.org
f960211f8b Fixes two jitter buffer bugs related to NACK.
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 16:00:49 +00:00
bjornv@webrtc.org
250cd6f41b Added a VAD unit test to common_audio. At this stage it runs through the API calls, but should later be complemented with tests on a file.
Review URL: http://webrtc-codereview.appspot.com/243002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@832 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 12:45:58 +00:00
stefan@webrtc.org
eb65860720 Reverts the workaround in r823 and solves a macro bug.
The macro bug caused frames to be dropped after being grabbed
for decoding.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 12:25:34 +00:00
tina.legrand@webrtc.org
8b1f621e3a Updated gypi for tests to work on osx.
Review URL: http://webrtc-codereview.appspot.com/250002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@830 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 08:57:34 +00:00
amyfong@webrtc.org
ca4666b75c vie wintest added hybrid protection mode
also fixed Max Framerate to reflect its actually the min framerate
Review URL: http://webrtc-codereview.appspot.com/244010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@828 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 21:16:40 +00:00
amyfong@webrtc.org
1e7e60b739 Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was
changed in r822.
Review URL: http://webrtc-codereview.appspot.com/240014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@827 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 20:53:30 +00:00
amyfong@webrtc.org
51e1bb4e1a vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll
When creating a new custom call, now able to set start bit rate (default is 1000)

The following modify call options were added
  9. Toggle Encoder Observer
 10. Toggle Decoder Observer
 12. Print Call Statistics

Also set the trace filter to kTraceAll

File defaults new call VGA (640x480)
Review URL: http://webrtc-codereview.appspot.com/239012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@826 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 18:16:35 +00:00
mikhal@webrtc.org
5200a05500 video_coding/jitter_buffer Updating condition on which we return a frame.
Review URL: http://webrtc-codereview.appspot.com/240011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:54:54 +00:00
mikhal@webrtc.org
30f6376802 VP8: Updating codec version: VP8 version will now return the libvpx version used.
Review URL: http://webrtc-codereview.appspot.com/247009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@824 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:45:00 +00:00
stefan@webrtc.org
2d28aff785 Workaround for an issue where frames are grabbed for decoding prematurely.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:13:18 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate.
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:08:29 +00:00
mflodman@webrtc.org
7e4269e9ee Changed VP8 qp min and added noise reduction.
Review URL: http://webrtc-codereview.appspot.com/248003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@821 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 12:59:47 +00:00
mflodman@webrtc.org
8fc663b3ae Don't trigger false ViE SetReceiveCodec warning.
Review URL: http://webrtc-codereview.appspot.com/250001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@820 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 11:30:52 +00:00
kjellander@webrtc.org
6b7799021c Fixing build errors on Windows platform. Minor changes...
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@819 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 02:38:09 +00:00
andrew@webrtc.org
fdde8b3fb7 Add references to src/ copies for LICENSE etc.
Review URL: http://webrtc-codereview.appspot.com/246007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@818 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 01:05:07 +00:00
andrew@webrtc.org
cb18121990 Add an unpacker tool for audioproc debug files.
- It only unpacks audio data at the moment.
- Also switch to Chrome's protoc.gypi for protobuf targets. This reduces
  the complexity of our targets.

Review URL: http://webrtc-codereview.appspot.com/241009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@817 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 00:27:17 +00:00
frkoenig@google.com
fc9bcef8c7 Data alignment fix for SSIM.
WebRtc_UWord64[2] wasn't always aligned to 128 bytes, which
is necessary for _mm_store_si128.  By declaring the 
variable as __m128i it will always be 128 bytes aligned.

Incorrect include files.

__m128i is defined in emmintrin.h for visual studio.  Extra include on mac and linux is not a problem.
Review URL: http://webrtc-codereview.appspot.com/239013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@816 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 00:07:32 +00:00
phoglund@webrtc.org
78c767f9ba Rewrote codec test to use fake camera.
Tests now fail more cleanly if the input video file is incorrect. Fixed some of the style issues in vie_autotest_codec.

Rewrote the automated standard codec test to use the new fake camera.

Started sketching on a new test case. Wrote a new abstraction called ViEFakeCamera which hides the details of how to thread a file capture device in the typical test case.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@815 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 12:54:38 +00:00
stefan@webrtc.org
d855c1a4e8 Reverts r807 and fixes the real issue in the VCM.
This fixes an issue in the VCM where we don't wait for a packet to arrive
if the jitter buffer is empty. This also fixes an issue where an old
packet can trigger a packet event signal for a future frame.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@814 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 11:52:48 +00:00
henrika@webrtc.org
bdb55c806f This CL is an attempt to remove a crash we can see when closing down VoiceEgine.
It can happen that the capture thread tries to access an invalid object after StopPlayout has been called.

I have also extended the usage of the new ScopedCOMInitializer to all threads. See this step as code cleanup.
Review URL: http://webrtc-codereview.appspot.com/239014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@813 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 11:03:28 +00:00
henrika@webrtc.org
a6c23357c0 Solves crash in ADM API unit test for Core Audio on Windows
Review URL: http://webrtc-codereview.appspot.com/244009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@812 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 08:31:33 +00:00
henrika@webrtc.org
5423bc2d0b Adds correct absolute paths to all input files in ADM functional unit tests.
Files are now read and played out correctly.
Review URL: http://webrtc-codereview.appspot.com/246006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@811 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 08:24:20 +00:00
kma@webrtc.org
ca325ececd Corrected a linux build error introduced in issue 246005.
Review URL: http://webrtc-codereview.appspot.com/246008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@809 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 02:36:09 +00:00
wjia@webrtc.org
f0cd394a2e Put fwrite calls under corresponding macros since they shouldn't show up in release build.
This also make chromeos build happy.
BUG=none
TEST=compile
Review URL: http://webrtc-codereview.appspot.com/247006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@808 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:40:43 +00:00
mikhal@webrtc.org
f31826e17b adding a wait on the decode thread when no frames are available
Review URL: http://webrtc-codereview.appspot.com/246009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@807 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:20:54 +00:00
mikhal@webrtc.org
a412924c0e VP8:Setting number of cores based on image size
Review URL: http://webrtc-codereview.appspot.com/242010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@806 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:02:43 +00:00
kma@webrtc.org
913644b92d For commiting changes in CL 277002, due to file structure changes introduced during
the review of the code.
Review URL: http://webrtc-codereview.appspot.com/246005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@805 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 21:36:33 +00:00
henrike@webrtc.org
0d0037c2fd Return cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac).
Review URL: http://webrtc-codereview.appspot.com/226005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@804 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:48:14 +00:00
phoglund@webrtc.org
0a9c318c9f The fread result is no longer ignored.
Changed unsigned longs into uint64_t to be a bit more portable.

Merge branch 'master' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc

Removed unnecessary use of WebRTC types. Fixed style issues.

Fixed style issues. Added comments where needed.

(After review) Made the standard base test not mirror the render stream since that is assumed to be tested in the render module. Renamed functions accordingly.

Fixed merge errors.
Merge branch 'master' into fake_camera

Conflicts:
	src/video_engine/main/interface/vie_capture.h
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/interface/vie_autotest_defines.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

More updates after review.

Updates after review.

Added new automated test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

Added comments to the new test.

- Fixed a bug which caused test error messages to not get shown.

- Added extended and API tests.
- Abstracted out an integration test base class since all integration
tests set up the exact same way.

- The ViETest::TestError static method will now assert using GTest
asserts if we are running in GTest mode. This gets rid of the hard
asserts that get run otherwise. The hard asserts are still in when using
"classic" mode. TestError will use neither GUnit nor hard asserts if
VIE_ASSERT_ERROR is not defined.
- Formatted vie_autotest_defines.h according to Google style rules.

- Extracted a method for finding a capture device on the system. This
removes a fair bit of logic from the huge test method (mostly straight
statements remain there now).

Rebase from svn.

- Whitespace fixes after review.

Fixed presubmit warning.

- Fixed cpplint.py warnings.

Fixed merge error.

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_extended_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/helpers/vie_window_creator.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

More updates after review.

Updates after review.

Added new automated test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

Added comments to the new test.

- Fixed a bug which caused test error messages to not get shown.

- Added extended and API tests.
- Abstracted out an integration test base class since all integration
tests set up the exact same way.

- The ViETest::TestError static method will now assert using GTest
asserts if we are running in GTest mode. This gets rid of the hard
asserts that get run otherwise. The hard asserts are still in when using
"classic" mode. TestError will use neither GUnit nor hard asserts if
VIE_ASSERT_ERROR is not defined.
- Formatted vie_autotest_defines.h according to Google style rules.

- Extracted a method for finding a capture device on the system. This
removes a fair bit of logic from the huge test method (mostly straight
statements remain there now).

Rebase from svn.

- Whitespace fixes after review.

Fixed presubmit warning.

- Fixed cpplint.py warnings.

Fixed merge error.

Fixed cpplint.py warnings.

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_api_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_extended_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_integration_test_base.cc
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/helpers/vie_window_creator.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_main.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

Revert "Revert "- Whitespace fixes after review.""

This reverts commit 3da2a148814e8dea78f73d3feeb32dce690dc2d4.

Revert "- Whitespace fixes after review."

This reverts commit fac670ca313580fb883191ae919091a2637ad0af.

- Whitespace fixes after review.

- Wrote a "file capture device" which is a kind of fake capture device. It reads a YUV file from disk and pretends that it is what the "camera" is seeing. This makes is possible to run tests based on video input without having an actual physical camera. This is good because physical cameras are quite unreliable. - Rewrote the standard mirrored preview loopback test so it can use the new file capture device. The old "classic" test is preserved. I tried to minimize duplication between the classic test case and the new one, which turned out to be quite painful. - There are some rough edges left in in the code. Suggested improvements is to get rid of the error counting mechanism since the code seems to assume that TestError invocations cause hard asserts anyway. The code will segfault for certain errors if the hard asserts doesn't happen, which means the error counting mechanism is unnecessary. This, by the way, could be a problem for the new test since it doesn't cause hard asserts. - Fixed comments for the thread wrapper and the external capture device interface.

- Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

- The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

- Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

- Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

- Fixed cpplint.py warnings.

Fixed presubmit warning.

- Whitespace fixes after review.

Rebase from svn.

- Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

- The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

- Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

- Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@803 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:33:07 +00:00
andrew@webrtc.org
537096a5c1 Remove unnecessary objective-c compiler flags.
Review URL: http://webrtc-codereview.appspot.com/239011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@802 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:07:08 +00:00
phoglund@webrtc.org
c63f788e0f Added fake camera, rewrote one test to use it.
Wrote a "file capture device" which is a kind of fake capture device. It reads a YUV file from disk and pretends that it is what the "camera" is seeing. This makes is possible to run tests based on video input without having an actual physical camera. This is good because physical cameras are quite unreliable.

Rewrote the standard mirrored preview loopback test so it can use the new file capture device. The old "classic" test is preserved. I tried to minimize duplication between the classic test case and the new one, which turned out to be quite painful.

There are some rough edges left in in the code. Suggested improvements is to get rid of the error counting mechanism since the code seems to assume that TestError invocations cause hard asserts anyway. The code will segfault for certain errors if the hard asserts doesn't happen, which means the error counting mechanism is unnecessary. This, by the way, could be a problem for the new test since it doesn't cause hard asserts.

Fixed comments for the thread wrapper and the external capture device interface.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/224003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@801 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 13:20:09 +00:00
henrika@webrtc.org
bf478faebb Ensures that ADM unit tests builds on all platforms.
Review URL: http://webrtc-codereview.appspot.com/240009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@800 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 10:31:02 +00:00
andrew@webrtc.org
f1a605cad6 Update DEPS to support Mac clang build.
Review URL: http://webrtc-codereview.appspot.com/244003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@797 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 15:29:16 +00:00
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 13:42:50 +00:00
stefan@webrtc.org
158f496030 Fixes a rate control bug in the VP8 wrapper.
Changes how we signal frame rate and frame durations to the encoder. Rather
than changing the time base, we now only modify the frame durations, while
keeping the timebase constant. The frame duration is currently calculated
from the average input frame rate. Ideally, the frame duration should
be calculated as the timestamp diff, which is the real duration of a
frame, but the encoder doesn't seem to like too varying durations.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@795 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 13:15:16 +00:00
stefan@webrtc.org
ead87b5051 Fix potential issue where frame buffers might be freed while being decoded.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/243004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@791 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 06:46:37 +00:00
stefan@webrtc.org
2b0f094c8f Avoid reallocating the decodedImage for every decoded frame.
Also made sure the right size is allocated.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@790 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 06:39:03 +00:00
mikhal@webrtc.org
ee3dfa6f43 Review URL: http://webrtc-codereview.appspot.com/241007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@789 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 00:46:09 +00:00
mikhal@webrtc.org
1af915d8ae video_coding: vp8: Updating error propagation threshold
Review URL: http://webrtc-codereview.appspot.com/246002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@788 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 18:19:18 +00:00
kma@webrtc.org
d75889e2eb Change of Android makefiles to build latest video coding code.
Review URL: http://webrtc-codereview.appspot.com/239008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@786 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 16:28:56 +00:00
henrika@webrtc.org
7cf893743a git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-10-20 12:30:35 +00:00
henrika@webrtc.org
cedbb036d1 [Issue 101] Solves memory leak on Windows
git-svn-id: http://webrtc.googlecode.com/svn/trunk@784 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 12:11:45 +00:00
stefan@webrtc.org
c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/241006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@782 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 08:19:34 +00:00
mflodman@webrtc.org
ae499a2ac8 Set correct codec info before sending frame to VCM.
Review URL: http://webrtc-codereview.appspot.com/240003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@780 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 05:55:46 +00:00
kjellander@webrtc.org
81f25f9ff8 Fixing build errors on Windows platform. Minor changes...
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@779 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 20:06:56 +00:00
wu@webrtc.org
f3f2f6abdb * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM.
* Split the WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER into WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE and WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER.
* Add DummyDeviceInfo for the case when WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE is not defined.
Review URL: http://webrtc-codereview.appspot.com/224005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@778 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:42:17 +00:00
henrike@webrtc.org
509c9c5d09 operator + is evaluated before ?:
Parenthesis ensures the intended behavior.
Review URL: http://webrtc-codereview.appspot.com/239003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@777 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:31:01 +00:00
henrike@webrtc.org
4df8c9a2ed Review URL: http://webrtc-codereview.appspot.com/243001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@776 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:30:25 +00:00
andrew@webrtc.org
7ecdf585cb Enable chromium_code:1 in the Chrome build.
Review URL: http://webrtc-codereview.appspot.com/240001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@775 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 17:53:56 +00:00
stefan@webrtc.org
ffd28f95c5 Request key frames to battle error propagation.
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).

For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 15:55:39 +00:00
mikhal@webrtc.org
d0752c370d video_coding: Update to hybrid mode: Set FEC values for zero below a threshold.
Review URL: http://webrtc-codereview.appspot.com/245001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@773 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 15:48:30 +00:00
mflodman@webrtc.org
c693bac6e7 Only start ViEPerformanceMonitor when needed.
Tested by taking the added part in base extended test and running in Standard test with cpu threashold in ViEPeroformanceMonitor manually changed to 0.

Review URL: http://webrtc-codereview.appspot.com/240005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@772 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 13:40:58 +00:00
phoglund@webrtc.org
b5475d0076 vie_auto_test will now obey the Mac .mm rules for files including objective-c code.
Fixed the Windows build.

Fixed whitespace.

Split the platform-specific code for creating a window manager into separate source files since the mac one must be suffixed .mm and not .cc when we happen to use objective-c code. Tested on Linux.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/214009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@771 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 10:59:39 +00:00
bjornv@webrtc.org
4c636764b7 Updated the AEC delay logging to output values in ms. PB output updated.
Review URL: http://webrtc-codereview.appspot.com/223003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@770 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 08:47:40 +00:00
mflodman@webrtc.org
cc412c1735 Remove second instance of ViE PerformanceMonitor.
Review URL: http://webrtc-codereview.appspot.com/244001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@769 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 08:27:30 +00:00
mflodman@webrtc.org
ce8813da4e Using id instead of name when setting Mac/QTKit capture device.
Review URL: http://webrtc-codereview.appspot.com/241002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@768 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 06:45:16 +00:00
andrew@webrtc.org
4d5d5c1267 Reorganize the audio_processing source.
- Remove main and source directories.
- Change .gyp, .gypi and Android.mk files correspondingly. No other
  source changes.

Review URL: http://webrtc-codereview.appspot.com/241001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@767 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 01:40:33 +00:00
andrew@webrtc.org
5d3bdf71ab Fix clang warnings in ViE autotest.
Review URL: http://webrtc-codereview.appspot.com/239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@766 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 01:09:41 +00:00
wu@webrtc.org
8fd93d4d96 Move DeliverCapturedFrame from private to protected.
Review URL: http://webrtc-codereview.appspot.com/246001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@765 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 00:16:36 +00:00
bjornv@webrtc.org
52eddf7378 Made Tina, Andrew and Jan as OWNERS to entire common_audio and removed the sub-OWNERS files. Let me know if that's fine.
Review URL: http://webrtc-codereview.appspot.com/225006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@763 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 07:57:04 +00:00
stefan@webrtc.org
5b15cfc6dd Fix BWE unit test build issue
git-svn-id: http://webrtc.googlecode.com/svn/trunk@762 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 07:22:33 +00:00
kjellander@webrtc.org
61f07c3184 I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files.
The ApmTest.Process test is still failing and needs to be resolved.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/194002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@761 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 06:54:58 +00:00