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Author SHA1 Message Date
minyue@webrtc.org
c1a40a7b68 This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. 
This CL is going to be combined with another CL in ACM, which is to be landed.

TEST=passed_try_bots
BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 09:52:06 +00:00
wu@webrtc.org
cb711f77d2 Add interface to propagate audio capture timestamp to the renderer. 
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:39:11 +00:00
henrika@webrtc.org
9f277350f8 Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12299005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 12:04:29 +00:00
henrika@webrtc.org
f383a1b0f2 Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 11:51:45 +00:00
henrika@webrtc.org
6b02eea6ac Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:24:10 +00:00
henrika@webrtc.org
1cec3957b8 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:19:19 +00:00
henrika@webrtc.org
66021e0fa2 Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 08:53:27 +00:00
henrika@webrtc.org
3b76627afe Removes parts of the webrtc::VoEHardware sub API (relanding) 
Relanding https://webrtc-codereview.appspot.com/18399004/

TBR=niklase

Review URL: https://webrtc-codereview.appspot.com/16489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:43:00 +00:00
henrika@webrtc.org
3106b706c0 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." 
> Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
> 
> BUG=3206
> R=andrew@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18399004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:10:50 +00:00
henrika@webrtc.org
9de3d844ae Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 10:55:11 +00:00
andrew@webrtc.org
8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>. 
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:10:28 +00:00
wu@webrtc.org
93fd25c20c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. 
* Cast rtp header extension to int in log in rtp_utility.cc.

BUG=3237
TEST=try bots
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 20:33:08 +00:00
henrika@webrtc.org
66803489f9 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=henrik.lundin@webrtc.org, juberti@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 10:45:01 +00:00
henrika@webrtc.org
0f7375504a Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=juberti@webrtc.org, niklas.enbom@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 10:38:08 +00:00
henrika@webrtc.org
b9309beea4 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 14:12:50 +00:00
henrika@webrtc.org
8883a0f47f (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds 
Landing https://webrtc-codereview.appspot.com/11419004/ manually.

TBR=niklase
BUG=none

Review URL: https://webrtc-codereview.appspot.com/11439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 13:04:12 +00:00
solenberg@webrtc.org
b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 10:38:25 +00:00
wu@webrtc.org
ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. 
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 23:49:08 +00:00
henrika@webrtc.org
b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. 
Reasons for removing:

- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.

BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 08:58:08 +00:00
solenberg@webrtc.org
a07923339b Remove external encryption API for VoE. 
BUG=
R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 11:27:22 +00:00
xians@webrtc.org
c1e28038ba Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-02 15:30:20 +00:00
xians@webrtc.org
07e5196414 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. 
The callback has to go through VoEBaseImpl since VoEChannel is internal to voice engine.

TEST=compile
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 13:54:02 +00:00
wu@webrtc.org
24301a67c6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 19:17:43 +00:00
henrikg@webrtc.org
863b536100 Allow opening an AEC dump from an existing file handle. 
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.

This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.

BUG=2567
R=andrew@webrtc.org, henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 16:05:17 +00:00
turaj@webrtc.org
03f33709f8 Inject config when creating channels to override the existing one. 
BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 00:02:48 +00:00
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