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Author SHA1 Message Date
jiayl@webrtc.org
bf00740c92 Adds a new voice engine warning for the typing noise off state. 
The old VE_TYPING_NOISE_WARNING is unchanged and fired whenever typing noise is detected.
The new VE_TYPING_NOISE_OFF_WARNING is fired when typing noise was detected and is gone now.
This is necessary for converting the typing state to a PeerConnection stats.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 18:09:20 +00:00
minyue@webrtc.org
e509f943ed This issue is related to 
https://chromereviews.googleplex.com/9908014/

I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.

BUG=
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:03:00 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:38:54 +00:00
pbos@webrtc.org
676ff1ed89 Ref-counted rewrite of ChannelManager. 
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.

ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.

BUG=2081
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1802004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-07 17:57:36 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 14:30:48 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
niklas.enbom@webrtc.org
b35d2e3abc Add dummy audio NACK APIs 
R=pwestin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-31 21:13:52 +00:00
turaj@webrtc.org
e46c8d3875 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. 
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:39:43 +00:00
pbos@webrtc.org
956aa7e087 Include files from webrtc/.. paths in voice_engine/ 
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 13:52:32 +00:00
pbos@webrtc.org
8a025e26db Make sure VoiceEngine tests only include one test framework. 
BUG=
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 11:25:12 +00:00
pbos@webrtc.org
9213521ea9 Remove const for plain data types in voice_engine/ 
BUG=1644
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1463004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 08:31:39 +00:00
pwestin@webrtc.org
1de01354e6 Adding playout buffer status to the voe video sync 
Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 20:23:35 +00:00
pbos@webrtc.org
6141e13873 WebRtc_Word32 -> int32_t in voice_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:09:10 +00:00
pwestin@webrtc.org
0c45957e3a Remove UDP transport API from VoE 
Review URL: https://webrtc-codereview.appspot.com/1236004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 15:43:57 +00:00
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +00:00
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +00:00
wu@webrtc.org
80fccc29de Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." 
> Removed CPU APIs from VoEHardware. Code is now only used by test applications.
> 
> BUG=8404677
> 
> Review URL: https://webrtc-codereview.appspot.com/1238004

TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 23:38:21 +00:00
henrike@webrtc.org
4c138e8fca Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
BUG=8404677

Review URL: https://webrtc-codereview.appspot.com/1238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 21:23:42 +00:00
andrew@webrtc.org
0633cccb4f Alphabetize include order in fake_voe_external_media.h. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 01:57:24 +00:00
andrew@webrtc.org
c83a00ad49 Add some VoE and AudioProcessing mocks. 
Includes a bit of shared helpers in fake_common.h.

Review URL: https://webrtc-codereview.appspot.com/1221004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 21:20:38 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665 
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. 
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 17:52:42 +00:00
turaj@webrtc.org
b7edd06530 Remove DTMF detection. Talk team has been in the loop and there is no need for 
DTMF detection at the receiver side.

test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 22:27:27 +00:00
turaj@webrtc.org
24045c5a02 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. 
bug=issue1370
test=trybots
Review URL: https://webrtc-codereview.appspot.com/1121007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 03:14:22 +00:00
andrew@webrtc.org
f0a90c37c4 Expose the capture-side AudioProcessing object and allow it to be injected. 
* Clean up the configuration code, including removing most of the weird defines.
* Add a unit test.

Review URL: https://webrtc-codereview.appspot.com/1152005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 01:12:49 +00:00
tommi@webrtc.org
0989fb7bfa Make VoiceEngineImpl inherit from VoiceEngine. 
This associates the two types instead of incorrectly reinterpret casting
VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated).

Please see more details in the bug for how this is currently causing problems
with security tools.

BUG=38612
Review URL: https://webrtc-codereview.appspot.com/1099013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 15:07:32 +00:00
turaj@webrtc.org
6388c3e2fd Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. 
TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 21:42:18 +00:00
roosa@google.com
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
roosa@google.com
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
turaj@webrtc.org
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
leozwang@webrtc.org
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
perkj@webrtc.org
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
turaj@webrtc.org
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
andrew@webrtc.org
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