Commit Graph

724 Commits

Author SHA1 Message Date
b2b628d5cd Further relax thresholds in mixing test.
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3327 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-02 18:50:13 +00:00
00c7c4315b Replace voice engine utility functions with system wrapper variants.
SLEEP -> SleepMs
GET_TIME_IN_MS -> TickTime::MillisecondTimestamp

These could cause unused function errors on some compilers.

BUG=1228

Review URL: https://webrtc-codereview.appspot.com/1013004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3326 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-02 16:06:39 +00:00
1c75918302 Disabled flaky test.
From flake in http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/270

Review URL: https://webrtc-codereview.appspot.com/1001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3293 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 10:40:05 +00:00
b8ba4d8109 Add number of inserted samples to NetEq statistics.
BUG=

Review URL: https://webrtc-codereview.appspot.com/964030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 00:06:18 +00:00
1b60ceb499 Add GetAudioFrame API to VoiceEngine.
Allows the caller to pull frames from a channel instead of sending them to the output mixer.

BUG=

Review URL: https://webrtc-codereview.appspot.com/973012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3273 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12 23:00:29 +00:00
b718619f0a Expose NetEq playout mode off through VoiceEngine.
BUG=

Review URL: https://webrtc-codereview.appspot.com/971016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12 21:59:14 +00:00
0870f02cdb Add API to retreive last received RTP timestamp to VoiceEngine.
BUG=

Review URL: https://webrtc-codereview.appspot.com/969016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3271 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12 21:31:41 +00:00
42259e7ebc VoE Changes to enable dual_streaming.
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Committed: https://code.google.com/p/webrtc/source/detail?r=3231
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3257 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-11 02:15:12 +00:00
96bcac8fbb Expose Set and Get Recording/Playout sample rate apis
Message:
This is the first cl to add Set/Get Recording and Playout sample rate apis.
In this cl, apis are enabled but returns -1, will add android
implementation in next cl, it's easy for review and coding.

Description:
This CL expose fours voice engine apis,
SetRecordingSampleRate,
RecordingSampleRate, 
SetPlayoutSampleRate,
PlayoutSampleRate. 

BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/626004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3239 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 19:11:55 +00:00
2cf22a6abc Revert 3231 - VoE Changes to enable dual_streaming.
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Review URL: https://webrtc-codereview.appspot.com/970005

TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929040

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3236 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 10:02:02 +00:00
767d87cf24 VoE Changes to enable dual_streaming.
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3231 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 22:51:37 +00:00
b0dff12d2b 48 kHz extension to iSAC.
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 17:43:52 +00:00
0f8286fd75 Added last (?) suppressions for known issues.
BUG=1152

Review URL: https://webrtc-codereview.appspot.com/933027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3180 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 14:21:12 +00:00
8d334d387b Disabled flaky test on Linux, added disable-on-platform macros, fixed \n's
BUG=1155

Review URL: https://webrtc-codereview.appspot.com/972006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3178 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 12:28:06 +00:00
52ec985d82 Fixing vie and voe auto test project paths for test execution.
By letting fileutils.h know the path to the executable, the tests will be able to find the project root dir and resource file paths even when the test is executed outside the checkout dir.

See http://review.webrtc.org/858014/ for more background.

Today, these tests are failing in the FYI waterfall since they are run "Chromium style" (i.e. from one level above the checkout dir). Since we're moving in that direction this needs to be fixed. It has been fixed for all other tests already.

TEST=Local test execution of vie_auto_test and voe_auto_test with CWD one level above trunk/

Review URL: https://webrtc-codereview.appspot.com/974004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3173 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-27 10:01:01 +00:00
655d8f56f6 Add a kTraceTerseInfo level for non-verbose logging.
Review URL: https://webrtc-codereview.appspot.com/937023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3134 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-20 07:34:45 +00:00
de727ab260 Fixes http://code.google.com/p/webrtc/issues/detail?id=941
BUG=941

Review URL: https://webrtc-codereview.appspot.com/966020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3125 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-18 18:49:13 +00:00
50419b0777 Add libjingle-style stream-style logging.
Add a highly stripped-down version of libjingle's base/logging.h. It is
a thin wrapper around WEBRTC_TRACE, maintaining the libjingle log
semantics to ease a transition to that format.

Also add some helper macros for easy API and function failure logging.

Review URL: https://webrtc-codereview.appspot.com/931010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3099 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-14 19:07:54 +00:00
d5fbdc8e52 Increase number of channels that can be supported on Android
BUG=
TEST=local
Review URL: https://webrtc-codereview.appspot.com/967005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3090 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13 21:30:34 +00:00
06d72d881f Add Android OWNER files
Message:
Add OWNER files so I can review and approve changes for Android.
I also should be owner for all .mk file, but it's OK for now,
please review.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/932016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3069 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 17:51:55 +00:00
c862f49b2d Move capture level computation after all processing.
BUG=issue1065

Review URL: https://webrtc-codereview.appspot.com/930014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3057 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-07 19:08:03 +00:00
ddcc9429e7 Check the channels in receive-side processing frames.
The number of channels must be set correctly before calling ProcessStream. This
was preventing stereo frames from being processed.

Also fix voe_cmd_test, which wasn't enabling rx NS properly.

BUG=issue713, 7375579

Review URL: https://webrtc-codereview.appspot.com/929013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3047 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-06 18:39:40 +00:00
512535097e Added buffer length when calling encrypt(). Write the extra two bytes.
BUG=934
TEST=Run VoE Autotest Encryption with Valgrind.
Review URL: https://webrtc-codereview.appspot.com/930004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2995 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25 13:58:02 +00:00
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00