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Author SHA1 Message Date
braveyao@webrtc.org
ab12990b1b In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately. 
Now we would get a failure if we do the same thing and the file would not be played out. Then GTalk/Hangout also reported this failure to us. 
This CL is to restore the original function. 

BUG = Issue 490
TEST = Manual test and voe_auto_test->FileBeforeStreamingTest
Review URL: https://webrtc-codereview.appspot.com/569016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 03:26:39 +00:00
tina.legrand@webrtc.org
4517585db5 Adding separate payload types for stereo modes 
BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 09:27:35 +00:00
andrew@webrtc.org
16fcb247b2 Disable flaky VolumeTests only on Linux. 
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/611005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 17:26:32 +00:00
andrew@webrtc.org
459955f821 Move audio_frame_operations to the utility module. 
TBR=henrika@webrtc.org
BUG=issue551
TEST=voe_auto_test, webrtc_utility_unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/599006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 22:13:14 +00:00
andrew@webrtc.org
aafa49bb85 Disable flaky VolumeTest.DefaultSpeakerVolumeIsAtMost255. 
This test failed on six CLs in a row recently.

TBR=xians@webrtc.org
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/595007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 22:05:15 +00:00
phoglund@webrtc.org
dbaa893525 Completed rewrite of APM extended test. 
Removed NS tests since they are already covered by audio_processing_test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/603004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 14:36:59 +00:00
leozwang@webrtc.org
351fb6d3b4 Exclude code that don't work on android in voe_cmd_test 
Description:
Ths cl makes voe_cmd_test work on android by excluding some code
that are availabel on android today, some highlights
1. change maxnumofchannles
2. disable audio device selection
3. disable set/get volume

BUG=
TEST=test on try bots
Review URL: https://webrtc-codereview.appspot.com/584009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-25 16:47:35 +00:00
andrew@webrtc.org
f45d47ad7d Remove mixing tests from voe_extended_test.cc 
These have been moved to:
src/voice_engine/main/test/auto_test/standard/mixing_test.cc

BUG=
TEST=build voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 17:27:59 +00:00
andrew@webrtc.org
51b4f3e6a8 Try to fix MixingTest on the Win bots. 
- Relax the constraints on recording duration.
- Remove unneeded file deletes. (These files will be properly
  overwritten anyway).

TBR=henrike@webrtc.org
BUG=issue534
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/600006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 17:26:05 +00:00
mflodman@webrtc.org
6af9594d71 Added gyp variable to include/exclude all tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 13:23:35 +00:00
niklas.enbom@webrtc.org
ee646c37d4 I know this is ugly, but it helps a lot to quickly update webRTC in Chrome and libJingle. 
Review URL: https://webrtc-codereview.appspot.com/596004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 11:41:02 +00:00
andrew@webrtc.org
7fbfc4ce79 Use correct variable in trace. 
TBR=leozwang@webrtc.org
TEST=build

Review URL: https://webrtc-codereview.appspot.com/593004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 22:22:36 +00:00
andrew@webrtc.org
9dc45dad1b Move trunk/test/data -> trunk/data 
BUG=
TEST=all trybot test failures passed locally

Review URL: https://webrtc-codereview.appspot.com/583007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 15:39:01 +00:00
andrew@webrtc.org
a1a34d675f Avoid flakiness by skipping more output verification. 
- Add a SCOPED_TRACE in case it flakes out again.
- The test doesn't need to be very long, so shorten it to save the bots some time.

TBR=henrike@webrtc.org
BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 00:45:00 +00:00
andrew@webrtc.org
294be77c2e Permit mixing mono and stereo streams. 
Add mixing tests based on older ones from the extended tests.

BUG=issue534
TEST=manual, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/576014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 03:28:41 +00:00
phoglund@webrtc.org
1ad477de3e Added a bit flip fuzz test to the voice engine. 
Extracted encryption classes to a new test library.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/564009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-18 08:02:37 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module. 
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 11:08:54 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 23:01:04 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 12:33:50 +00:00
andrew@webrtc.org
270e9db039 Clarify the impact of disabling VAD on DTX. 
TBR=henrika@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/566009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 19:09:03 +00:00
niklas.enbom@webrtc.org
f6edfeff63 Adding one parameter to typing detection tuning 
Review URL: https://webrtc-codereview.appspot.com/569009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 13:16:12 +00:00
andrew@webrtc.org
e59a0aca6a Fix AudioFrame types. 
volume_ is not set anywhere so I'm removing it.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 17:12:40 +00:00
andrew@webrtc.org
589673f1cb Fix volume setting while not playing on PulseAudio. 
We now only set the volume when starting playout if the user has called
SetSpeakerVolume while we weren't playing. This now also ensures it will
actually get set to what the user requested rather than being overridden
by a default value.

Add tests to voe_auto_test.

BUG=6140661
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/566006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 21:42:49 +00:00
braveyao@webrtc.org
ba0f9fe10b Trival fix to voe_auto_test according to the main source codes 
BUG = NULL
TEST = voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/554004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 10:06:43 +00:00
andrew@webrtc.org
63a509858d Rename AudioFrame members. 
BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/542005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 23:56:37 +00:00
phoglund@webrtc.org
719dba7e79 Further cleaned up voe_standard_test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/522003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 07:32:37 +00:00
andrew@webrtc.org
a88cb6fce0 Add HighPassFilter and StereoChannelSwapping tests. 
BUG=issue451
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/531001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 17:00:20 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
andrew@webrtc.org
9c4f6a5ff9 Add an AudioFrameOperations unittest. 
Additionally, reformat audio_frame_operations to Goog style.

BUG=issue451
TEST=voice_engine_unittests

Review URL: https://webrtc-codereview.appspot.com/528001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 22:32:03 +00:00
tommi@webrtc.org
a990e122c4 * Change the reference counting implementation for VoE to be per object and 
not per interface. This simplifies things a bit, reduces code and makes it
  possible to implement reference counting (if we ever want) without the
  static Delete() method.  (Reference counted objects are traditionally
  implicitly deleted via the last Release())

* Since the reference counting code is now simpler, there's no need for the
  RefCount class so I'm removing it.

* Also removing the optional |ignoreRefCounters| variable from the VoiceEngine::Delete
  method.  The justification is that it's no longer used and the reason it was there
  in the first place was to avoid bugs in third party code, so it's an indication that
  something is wrong elsewhere.

* Fix a crash in voe_network_impl and voe_extended_test that would happen on machines with IPv6 support.

* Fix an assert (handle the case) in the extended audio tests when SetCNAME is called with a NULL pointer.

* As a side effect of this change, hopefully the footprint of VoE will be slightly smaller :)

BUG=10445 (Coverity)
Review URL: https://webrtc-codereview.appspot.com/521003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 15:28:22 +00:00
andrew@webrtc.org
1c7bfe02f7 Fail silently when swapping mono. 
TBR=tina.legrand@webrtc.org
BUG=issue451
TEST=forthcoming unittest

Review URL: https://webrtc-codereview.appspot.com/527003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 00:20:28 +00:00
andrew@webrtc.org
02d7174722 Add API to swap stereo channels. 
BUG=issue451
TEST=manually with voe_cmd_test, using stereo and mono codecs

Review URL: https://webrtc-codereview.appspot.com/519001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 19:47:00 +00:00
andrew@webrtc.org
369166a179 Add API for disabling the high pass filter. 
BUG=issue419
TEST=manually with voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/509003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 18:38:03 +00:00
leozwang@webrtc.org
48a5df6481 Embed svn revision number into code 
BUG=
TEST=build on linux
Review URL: https://webrtc-codereview.appspot.com/516001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 14:50:50 +00:00
phoglund@webrtc.org
b73f01e7fd Removed some obviously dead stuff from voe_auto_test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/495001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 10:59:31 +00:00
phoglund@webrtc.org
a36a4bb340 Disabled flaky voe tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/491007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-13 10:11:15 +00:00
tina.legrand@webrtc.org
16b6b90a82 Split of stereo packets moved 
In this CL I have rewritten the way we handle stereo packets in VoE.
Before this change we split the packets in the RTP module and added two packets to ACM, one for the left channel and one for the right. This lead to timing problems caused when a different thread called RecOut in between the two calls to add stereo packet to ACM. (RecOut is called to pull audio data, decode packets, on the receiving side).

While doing the change I also took the opportunity to changed some functions so that the data stream is uint8 everywhere.

The list of files in this CL is long, but should be fairly easy to review. It is difficult to see what has been changed  in some of the tests, but I can explain offline.

Reviewers:
Björn - /src/modules/audio_coding
Patrik - /src/modules/rtp_rtcp
Patrik -/src/modules/utility
Henrik A - /src/voice_engine

BUG=410
TEST=voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/473003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-12 11:02:38 +00:00
phoglund@webrtc.org
e1bbdb488e Rewrote external media test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/482002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-11 14:15:48 +00:00
andrew@webrtc.org
e713fd0eee Enable the "unused variable" warning on Windows. 
- Break out direct_show_base_classes to its own gyp file to have it
  treated as third party code.
- Fix the resulting warnings (courtesy of Tommi).

BUG=
TEST=build on Windows (vie_auto_test currently failing at HEAD)

Review URL: https://webrtc-codereview.appspot.com/489001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-10 07:13:46 +00:00
phoglund@webrtc.org
a1facdcf0f Re-enabled video sync tests (new attempt). 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/478001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-05 16:59:01 +00:00
tommi@webrtc.org
851becd00c Remove public virtual voe::SharedData inheritance. 
This is a fix for coverity issues: 10446, 10445, 10444, 10443.

Although the cl is rather big, there aren't many code changes:

* Instead of an implicit vtable pointer, there is now an explicit |_data| member to access the shared data.
* We don't access the member variables of SharedData directly.  There are accessors instead.
* SharedData setters that set values that must be freed, automatically free the previous value and 'addref' if required the new one.
* Lots and lots of 'rewrapping' due to search/replace after the above changes.

BUG=10446, 10445, 10444, 10443
TEST=Run all tests for VoE.
Review URL: https://webrtc-codereview.appspot.com/472009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-04 14:57:19 +00:00
niklas.enbom@webrtc.org
06e722ae77 Adding parameter setting for typing detection 
Review URL: https://webrtc-codereview.appspot.com/476001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-04 07:44:27 +00:00
phoglund@webrtc.org
afc39731dc Rewrote NetEQ stats test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/466002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-03 23:09:42 +00:00
phoglund@webrtc.org
f3bbc3e5b3 Temporarily disabled new test since it segfaults randomly. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/474002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 23:13:33 +00:00
phoglund@webrtc.org
9b96e02c20 Adjusted the deviation limit since the test seems to fail on the bot. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/471002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 22:20:48 +00:00
phoglund@webrtc.org
b5617869fc Fixed problem with previous commit. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/472002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 21:02:45 +00:00
phoglund@webrtc.org
e5f74bdbbc Rewrote the video sync test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/463001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 19:51:11 +00:00
niklas.enbom@webrtc.org
3dc886561c Adding time since last typing 
Review URL: https://webrtc-codereview.appspot.com/471001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 09:53:54 +00:00
phoglund@webrtc.org
1b1a39fdef Rewrote external encryption test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/456009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-29 16:58:35 +00:00
xians@webrtc.org
f35f54bf68 Fix coverity warning. 
Review URL: https://webrtc-codereview.appspot.com/465002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-29 13:59:24 +00:00