generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
2,340 Commits 1 Branch 0 Tags 108 MiB
2d4c4ae553
Commit Graph

225 Commits

Author SHA1 Message Date
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
braveyao@webrtc.org
d713143d99 To support playing mono file with stereo codec as mixing with microphone capture 
BUG=413
TEST=Manual test.
Review URL: https://webrtc-codereview.appspot.com/460004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-29 10:39:44 +00:00
niklas.enbom@webrtc.org
40197d7b3b Fixing build issus on non-Win 
TBR: bjornv
Review URL: https://webrtc-codereview.appspot.com/460005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-26 08:45:47 +00:00
niklas.enbom@webrtc.org
5398d9583b Force commit of 449006' 
Review URL: https://webrtc-codereview.appspot.com/455006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-26 08:11:25 +00:00
mflodman@webrtc.org
3e820e5109 Remove RTP Keep-alive from VoE and ViE. The RTP module functionality will be removed in a follow-up CL shortly. 
TEST=VoE autotest and ViE autotest

Review URL: https://webrtc-codereview.appspot.com/458002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-23 09:41:44 +00:00
phoglund@webrtc.org
4aa57b4150 Extracted a helper library from vie_auto_test. 
This CL does not attempt to fix the style issues in the moved tb_ files, at least not yet. In general I've tried to avoid dependencies between the library and vie_auto_test: vie_auto_test depends on the library but not the other way around. I had to make some slight changes to achieve this. I had to remove some ViETest::Log statements in tb_interfaces.cc and I had to move knowledge of where to put output files to the library. I think it ended up being pretty clean in the end but let me know if I missed something. I tried to convert all paths I touched to src-relative paths, so look out if I missed something there.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/450004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1923 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-22 12:56:54 +00:00
phoglund@webrtc.org
aaf62ac019 Temporarily disabled flaky tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/446010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1919 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-21 08:04:43 +00:00
phoglund@webrtc.org
7e26ad3828 Disabled more flaky volume tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/451012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-16 09:46:52 +00:00
henrika@webrtc.org
907bc55c19 Removes WebRtc_Word8 dependecy in the AudioDeviceModule. 
This CL also modifies the ADM callback interface and introduces void* instead of WebRtc_Word8*
as pointer types for data buffers. This change also affects the VoiceEngine.
Review URL: https://webrtc-codereview.appspot.com/443001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-09 08:59:19 +00:00
andrew@webrtc.org
8012474552 Use a const rather than macro for EcDefault. 
- This should be a better solution to the build error in
  https://webrtc-codereview.appspot.com/425005
- Ideally all of the similar macros should go away, but one thing at
  a time...

BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/438002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-08 17:54:24 +00:00
mflodman@webrtc.org
9a065d1eae VoiceEngine now uses pointer constructor of CriticalSectionScoped, instead of reference. 
BUG=184
TEST=Compiles on all platforms.

Review URL: https://webrtc-codereview.appspot.com/436001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-07 08:12:21 +00:00
leozwang@webrtc.org
30185916aa Fix error in test app which was introduced when payload type was converted to int 
TBR=mflodman, phoglund
Review URL: https://webrtc-codereview.appspot.com/439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 21:43:55 +00:00
leozwang@webrtc.org
0975d2158c Cleanup messy data type of unknown_payload_type 
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 20:59:13 +00:00
andrew@webrtc.org
8b111eb3e6 Reformat voe_audio_processing_impl to Goog style. 
TBR=xians@webrtc.org
BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/439003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 19:50:12 +00:00
andrew@webrtc.org
6f9f817e06 Add an API to offset system delay. 
Plumb it through VoiceEngine.

BUG=
TEST=voe_auto_test,audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/428010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 19:03:39 +00:00
leozwang@webrtc.org
53ed5a41a2 Fix building errors and enable test app 
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/428008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 20:15:58 +00:00
leozwang@webrtc.org
e47efe291e Fix building error on android 
Review URL: https://webrtc-codereview.appspot.com/425005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 16:05:30 +00:00
andrew@webrtc.org
d62d7301f4 Remove TARGET_PC and cruft from typedefs.h. 
Additionally remove all TARGET defines (e.g. TARGET_MAC), which weren't used anyway.

BUG=
TEST=build on Linux, Mac, Win

Review URL: https://webrtc-codereview.appspot.com/432001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 21:39:57 +00:00
andrew@webrtc.org
fa2f5627ca Change error code. 
TBR=henrika@webrtc.org
BUG=
TEST=build

Review URL: https://webrtc-codereview.appspot.com/429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 21:19:01 +00:00
leozwang@webrtc.org
813e4b0af0 Correct WebRtc_Word8 in voice engine 
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/425002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 18:34:25 +00:00
andrew@webrtc.org
0e28566247 Only reset AudioProcessing if number of channels has changed. 
Calling SetSendCodec() would always reset AudioProcessing.

BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/417002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 17:00:56 +00:00
leozwang@webrtc.org
07c68b9c9d Correct wrong usage of WebRtc_Word8 in rtp and udp module 
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/418001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 16:09:51 +00:00
phoglund@webrtc.org
2d124f3d88 Enabled the volume tests we believe are nonflaky and the vie_auto_test extended tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/422002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 14:34:06 +00:00
phoglund@webrtc.org
b45ceed9ef Rewrote the call report test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/399006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:55:04 +00:00
leozwang@webrtc.org
a52838b684 Update Android.mk and add test app 
Review URL: https://webrtc-codereview.appspot.com/388010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 01:16:43 +00:00
xians@webrtc.org
3ab6dda5cb Truncated the volume to 255 when the users set the volume above 100%. 
Allowed the users to set the volume above 100% when AGC is enabled, in this case AGC can gradually scale down the volume instead of jumping to 100% immediately.
Reduced the flakiness of the volume tests in linux.
Review URL: https://webrtc-codereview.appspot.com/387011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 18:15:54 +00:00
braveyao@webrtc.org
59d6cec291 Fix the crash at playing 48kHz stereo wav file. 
http://code.google.com/p/webrtc/issues/detail?id=208
Review URL: https://webrtc-codereview.appspot.com/396001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 18:17:16 +00:00
phoglund@webrtc.org
292da24166 New attempt. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 15:21:33 +00:00
phoglund@webrtc.org
dbe1e13b53 Fixed compilation error on Windows. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 14:03:44 +00:00
phoglund@webrtc.org
6b3bb89f12 Rewrote file test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:14:54 +00:00
phoglund@webrtc.org
aaa76f3ba8 Rewrote network test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/383003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 16:41:30 +00:00
phoglund@webrtc.org
78088c2f36 Removed warnings on Windows and enabled warnings-as-errors on Windows. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:56:45 +00:00
niklas.enbom@webrtc.org
87885e8409 This CL will look a bit strange. Essentially I've removed sanity checks for > 1 channel and then fixed the bugs that remained. Will add testing in a separate CL. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/390001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:48:59 +00:00
wu@webrtc.org
06c7dbae14 Disable flaky test AudioProcessingTest.TestVoiceActivityDetectionWithObserver. 
BUG=263
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/380009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 23:13:21 +00:00
phoglund@webrtc.org
56b85c6ba8 Reduced potential for flakiness in voice detection tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 18:48:33 +00:00
mflodman@webrtc.org
c80d9d9361 Removed default cases causing clang errors, -Wcovered-switch-default. 
BUG=
TEST=Bulid with clang version 3.1 (trunk 148911)

Review URL: https://webrtc-codereview.appspot.com/379008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 10:11:25 +00:00
kma@webrtc.org
de66b91274 In voice engine, added member audioFrame to classes AudioCodingModuleImpl and VoEBaseImpl, 
and touched VoEBaseImpl::NeedMorePlayData and AudioCodingModuleImpl::PlayoutData10Ms(), for
performance reasons in Android platforms.
The two functions used about 6% of VoE originally. After the change, the percentage reduced
to about 0.2%.
Review URL: https://webrtc-codereview.appspot.com/379001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-01 18:39:44 +00:00
xians@webrtc.org
79af734807 This patch fixes the converity warnings in voice engine. 
Review URL: https://webrtc-codereview.appspot.com/373017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 12:22:14 +00:00
henrika@webrtc.org
2919e95c2a Resolves Coverty issue #10347. 
Uninitialized member (UNINIT_CTOR).
Review URL: https://webrtc-codereview.appspot.com/369023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 08:45:03 +00:00
phoglund@webrtc.org
048eb7cda6 Finished rewriting the audio processing test. 
Partial rewrite of audio processing tests.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 11:58:41 +00:00
andrew@webrtc.org
b9d7d934de Rename interface/ to include/ in audio_processing. 
BUG=none
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/367007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 19:21:13 +00:00
andrew@webrtc.org
24bd58e689 Properly count anonymous mixing participants. 
When _amountOfMixableParticipants == 1, we skip mixing and saturation
protection. Without this fix, an anonymous participant would only be
properly counted if it was the last added.

For example, if an anonymous participant was added first, followed by
a regular participant, _amoutOfMixableParticipants would == 1 and the
regular participant would not be mixed.

BUG=issue209
TEST=New test added to voe_auto_test to verify, and used voe_cmd_test.

Review URL: https://webrtc-codereview.appspot.com/367006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 18:57:44 +00:00
andrew@webrtc.org
eeaf3d1fc1 Merge /branches/3.2:r1380 to /trunk 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 06:30:02 +00:00
leozwang@webrtc.org
f5cacdce8c Fix line aligement 
Review URL: https://webrtc-codereview.appspot.com/373002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 23:14:13 +00:00
phoglund@webrtc.org
12dbc23851 Rewrote volume test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:03:04 +00:00
phoglund@webrtc.org
3b57ee0238 Rewrote DTMF test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/368001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 09:22:33 +00:00
leozwang@webrtc.org
2638577f03 Add an argument in ANDROID_NOT_SUPPORT macro 
Review URL: https://webrtc-codereview.appspot.com/363003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 18:45:45 +00:00
tommi@webrtc.org
9ff87db5c0 Remove the diamond inheritance pattern from VoEVideoSyncImpl in attempt to see if this fixes coverity reports. 
CID=10446,10445,10444,10443
Review URL: https://webrtc-codereview.appspot.com/343018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:05:36 +00:00
punyabrata@webrtc.org
ad1927d368 Changing the typing detection sensitivity as the current 
setting does not work well in some scenarios especially
using webcams with built-in microphones.
Review URL: https://webrtc-codereview.appspot.com/349009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 18:53:04 +00:00
phoglund@webrtc.org
5badc7e969 Put system cpu tests back in, improved documentation. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/350011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 10:46:07 +00:00
phoglund@webrtc.org
c12f815de6 Rewrote hardware test and fixed broken tests on Windows. 
Fixed broken tests on Windows, including old tests.

Rewrote hardware test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/347008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 12:40:18 +00:00
henrika@webrtc.org
f75901fa4c Resolves CID 10540: Copy into fixed size buffer (STRING_OVERFLOW). 
You might overrun the 32 byte fixed-size string "receiveCodec.plname" by copying "payloadName" without checking the length.
Note: This defect has an elevated risk because the source argument is a parameter of the current function.
Review URL: http://webrtc-codereview.appspot.com/352009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 08:45:42 +00:00
braveyao@webrtc.org
f5c6573725 fix defect http://code.google.com/p/webrtc/issues/detail?id=215, audio device is not stopped appropriately. 
Review URL: http://webrtc-codereview.appspot.com/350008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 03:04:46 +00:00
andrew@webrtc.org
7859e10985 Propagate decoding errors to the mixer module. 
Review URL: http://webrtc-codereview.appspot.com/348001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 00:30:11 +00:00
henrik.lundin@webrtc.org
053c7991e3 Add minimum waiting time to NetEQ metrics 
Adding minWaitingTimeMs to ACMNetworkStatistics and to
NetworkStatistics. Also adding unittest.

TEST=audio_coding_unittests

Review URL: http://webrtc-codereview.appspot.com/350006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 14:16:44 +00:00
kjellander@webrtc.org
7f3c724e12 Renaming 47 files from .cpp to .cc 
In addition to our naming guidelines, this will cause these files to get parsed by Sonar, and to make searching/grepping the source using file extensions easier in the future.

BUG=
TEST=Compiling on Linux.

Review URL: http://webrtc-codereview.appspot.com/348005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 10:23:41 +00:00
perkj@webrtc.org
ce5990cb0b Fix defect http://code.google.com/p/webrtc/issues/detail?id=222 
"ViE GetSentRTCPStatistics fails on a sending channel if it don't receive rtp video packets.

BUG=222
TEST= tested in loopback. No new test added yet.

Review URL: http://webrtc-codereview.appspot.com/343003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 13:00:08 +00:00
phoglund@webrtc.org
01530a2ac2 Rewrote the rcp_rtcp test. 
Finished rewriting the rtp_rtcp test.

Rewrote first RTP RTCP test

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/342007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 12:26:34 +00:00
phoglund@webrtc.org
0aa7b32652 Finished rewriting the codec test. 
Rewrote more tests.

Rewrote most of the codec test and removed it from the regular test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-09 11:15:46 +00:00
pwestin@webrtc.org
c450a19669 Removed Version function from all modules. 
TBR=henrik_a
Review URL: http://webrtc-codereview.appspot.com/329023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 15:00:12 +00:00
henrik.lundin@webrtc.org
d439870473 Adding two new network metrics to NetEQ 
Clock-drift (in parts-per-million) and peaky-jitter mode status.
Both metrics are propagated to the VoE API. Tests are added
in the NetEQ unittests, and to some extent in ACM unittests
and VoE tests.

Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.

Note: The file neteq_network_stats.dat in resources must be updated
for the unittests to pass.

Review URL: http://webrtc-codereview.appspot.com/337005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 13:09:55 +00:00
andrew@webrtc.org
3192d655bd Fix for devices lacking stereo support. 
The number of capture channels can only be determined upon receiving the
first captured frame. We now assume stereo capture by default and set the
number of AudioProcessing input channels based on captured frames.

TEST=Windows mono-only device now runs AudioProcessing correctly (NS etc.), voe_auto_test (though some new, seemingly unrelated, tests are failing)

Review URL: http://webrtc-codereview.appspot.com/330013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 18:00:59 +00:00
henrik.lundin@webrtc.org
dbba1f969f Packet waiting-time statistics 
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.

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

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

Unit tests are also added for NetEQ and AcmNetEq.

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:45:05 +00:00
phoglund@webrtc.org
f3cea2336b Added an empty voice engine unit test binary in order to get correct coverage measurements. This will make the voice engine show up in the coverage measurements. The empty test is necessary to get the coverage tool to pick it up (and it will be easier to start writing unit tests for the voice engine later). 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/334003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 10:14:53 +00:00
phoglund@webrtc.org
fda17c2b00 Rewrote NetEQ test, made standard suite run googletestified tests too. 
The standard suite will now also run the googletestified tests.

Removed NetEQ tests from the standard test.

Initial version of new neteq test. Moved fixtures to own folder.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:07:37 +00:00
phoglund@webrtc.org
86a9f9b946 Fixed build error. 
Merge branch 'master' into voe_rewrites

Conflicts:
	src/voice_engine/main/test/auto_test/standard/after_streaming_fixture.cc

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Fixed strange build error.

Merge branch 'master' into voe_rewrites

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Nit fixes

Clarified some comments and method names.

Style fixes.

Removed tab characters.

Merge branch 'master' into voe_rewrites

Conflicts:
	src/voice_engine/main/test/auto_test/voe_standard_test.cc

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

Rewrote the hold test.

Abstracted out resource handling and created a new fixture for starting and stopping playing.

Rewrote network-before-streaming.

Revert "Rewrote network-before-streaming."

This reverts commit f1a07b813a90e4feef0a0737ebde9fb15acfd459.

Rewrote network-before-streaming.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 10:24:46 +00:00
phoglund@webrtc.org
188fc35e07 Rewrote the hold and netw-before-streaming tests. 
Rewrote the hold test.

Abstracted out resource handling and created a new fixture for starting and stopping playing.

Rewrote network-before-streaming.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/331001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 09:36:03 +00:00
phoglund@webrtc.org
610e90e910 Completed rewrite of codec test. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 10:40:19 +00:00
leozwang@webrtc.org
eda2da796e Fix compilation errors 
Review URL: http://webrtc-codereview.appspot.com/322014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 20:03:09 +00:00
phoglund@webrtc.org
667eca6290 Rewrote the hardware-before-streaming test. 
Restructured the test hierarchy somewhat - there is now a fixture for before-voe-init time and one for after-voe-init time.

Rewrote the hardware-before-streaming test.
Separated unrelated tests out from the rtp_rtcp tests.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 13:55:34 +00:00
phoglund@webrtc.org
fe61bc3607 Merge branch 'master' into voe_create_test 
Fixed broken build.

Nit fix.

Fixed style issues.

Removed accidental comment-out.

Removed test that no longer makes sense.

Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 17:02:16 +00:00
phoglund@webrtc.org
6418a24795 Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/322003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 16:24:23 +00:00
phoglund@webrtc.org
dd094fd6ae Started extracting methods out of the main test. 
Started extracting methods out of the main test, which will hopefully make us able to make the tests independent.

Merge branch 'master' into voe_split_methods

Conflicts:
	src/voice_engine/main/test/auto_test/voe_extended_test.cc
	src/voice_engine/main/test/auto_test/voe_extended_test.h
	src/voice_engine/main/test/auto_test/voe_standard_test.cc
	src/voice_engine/main/test/auto_test/voe_standard_test.h

Extracted methods out of the standard test.

Added space before inheritance colons.

Rolled back some header file changes.

Fixed long lines.

Fixed long lines.

Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/313001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 15:07:59 +00:00
phoglund@webrtc.org
693240f2d9 Fixed many formatting and indentation problems in voe_auto_test. 
Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 12:32:58 +00:00
henrika@webrtc.org
af71f0e5d9 Fixes two minor issues reported by the Coverty Integration Manager. 
BUG=none
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/302002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 07:02:22 +00:00
perkj@webrtc.org
68f2168978 Remove global voe::Channel::numSocketThreads. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:11:23 +00:00
henrik.lundin@webrtc.org
524eb48081 Removing deprecated NetEQ APIs 
Removing WebRtcNetEQ_GetPreferredBufferSize and
WebRtcNetEQ_GetCurrentDelay and all dependent APIs.

Review URL: http://webrtc-codereview.appspot.com/289006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 16:21:22 +00:00
xians@webrtc.org
e07247af8d Valgrind reports a racing condition on _sending because it is accessed by 
both TransmitMixer::PrepareDemux() and StartSend()/StopSend().
Put a lock to resolve it.
Review URL: http://webrtc-codereview.appspot.com/293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 16:31:28 +00:00
xians@webrtc.org
83661f534e fixing the racing conditions 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 10:58:15 +00:00
braveyao@webrtc.org
0a18522e1b Add support to 96kHz sampling rate to Windows CoreAudio interface. 
Review URL: http://webrtc-codereview.appspot.com/295003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 02:45:39 +00:00
henrik.lundin@webrtc.org
df10de4b27 Removing statistics API from NetEQ 
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.

Review URL: http://webrtc-codereview.appspot.com/285002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 09:36:23 +00:00
henrike@webrtc.org
31d30700d6 Addressed review comments from http://webrtc-codereview.appspot.com/256004/ 
Review URL: http://webrtc-codereview.appspot.com/256007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 19:59:32 +00:00
kjellander@webrtc.org
3f1cb8e546 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:56:54 +00:00