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36 Commits

Author SHA1 Message Date
Peter Kasting
6955870806 Convert channel counts to size_t. 
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
 2016-01-13 00:26:55 +00:00
Henrik Kjellander
1323fc39ba Remove webrtc/test/channel_transport/include 
Move the header file into webrtc/test/channel_transport instead.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel
R=henrika@webrtc.org, henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1431983006 .

Cr-Commit-Position: refs/heads/master@{#10595}
 2015-11-11 09:34:35 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}. 
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
 2015-09-25 11:58:39 +00:00
ivoc
b04965ccf8 Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. 
An option was added to voe_cmd_test to make a RtcEventLog dump.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1267683002

Cr-Commit-Position: refs/heads/master@{#9901}
 2015-09-09 07:09:49 +00:00
Minyue Li
79c143312b Delete VoiceChannelTransport before deleting Channel in voe_cmd_test 
Current voe_cmd_test shows following error when quitting:
DeRegisterExternalTransport() failed to locate channel.

This is to fix it.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45349004

Cr-Commit-Position: refs/heads/master@{#9129}
 2015-05-04 09:21:00 +00:00
Ivo Creusen
adf89b7e33 Added SetBitRate function to VoE API to allow changing the audio bitrate. 
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50789004

Cr-Commit-Position: refs/heads/master@{#9115}
 2015-04-29 14:03:45 +00:00
minyue@webrtc.org
9b2e1144df Supporting Opus DTX in Voice Engine. 
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.

BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43709004

Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 09:38:55 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
bjornv@webrtc.org
63da1dd972 audio_processing: Now records mic volume level also when using new AGC 
Previously only mic level calculated by the legacy agc was logged in aecdebug dumps.
Now we log it for any agc.
In addition, it is now possible to turn on and off debug recording in the test tool voe_cmd_test.

BUG=4274
TESTED=verified using voe_cmd_test
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39839004

Cr-Commit-Position: refs/heads/master@{#8274}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 19:44:46 +00:00
minyue@webrtc.org
456f01441a Re-allowing RED in voice engine. 
Path of audio RED packets was blocked in r4692 by accident. It ought be enabled again.

BUG=3619
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 11:58:42 +00:00
minyue@webrtc.org
adee8f9242 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate 
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 12:28:06 +00:00
pbos@webrtc.org
047a46f8b4 Remove Android.mk build files. 
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 08:48:51 +00:00
minyue@webrtc.org
4521e2d0bd Adding online bitrate change to voe_cmd_test 
This is to verify a way of changing the bitrate on-the-fly under current WebRTC implementation.

TEST=changing bit rate for different codecs. sound quality changed when bit rate was set successful. catched error when bit rate is invalid for a running codec.

BUG=
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 12:15:27 +00:00
minyue@webrtc.org
6aac93bd9c Adding SetOpusMaxBandwidth in VoE and ACM 
This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.

TEST = added a test in voe_cmd_test and listened to the result

BUG=
R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 08:13:33 +00:00
minyue@webrtc.org
2a8df7c375 Fixing two bugs in voe_cmd_test. 
I am trying to add a new functionality to voe_cmd_test, and I found two bugs:

1. in r5928, a functionality was removed but the item in the menu was not. Functionalities after it are offset.

r5928: https://code.google.com/p/webrtc/source/detail?r=5928&path=/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

2. in r6736, opus are set to output 48 kHz audio. When mixing Opus output with an audio file, channel.cc may go wrong.

r6736: https://code.google.com/p/webrtc/source/detail?r=6736

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 10:05:19 +00:00
aluebs@webrtc.org
9825afc3bd Add ExperimentalNs support in Config 
R=andrew@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 17:39:53 +00:00
henrik.lundin@webrtc.org
26e2b687fc Remove ACM1/ACM2 switching from VoiceEngine tests 
The option to run VoiceEngine tests with both ACM1 and ACM2 was
introduced while the two versions of AudioCoding module where both
in use. Now, ACM1 is being deprecated, and the tests should use the
defualt one (ACM2).

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 08:39:41 +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
andrew@webrtc.org
19018ddb17 Make ACM2 the default in voe_cmd_test. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 20:58:05 +00:00
fischman@webrtc.org
a789f3720a VoiceEngine(iOS & Android): removed NOT_SUPPORTED 
Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
  bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds

BUG=2050,3132
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 00:16:35 +00:00
henrika@webrtc.org
800b8dbda6 Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. 
BUG=none
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-20 08:07:41 +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
henrikg@webrtc.org
c693704cc2 Move out typing detection to its own class. 
This will allow an embedder to use it directly.

Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)

R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 09:50:46 +00:00
andrew@webrtc.org
7de3bb9df9 Output logs to stderr from voe_cmd_test by default. 
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.

TESTED=logs output to stderr by default and to the usual file when the
flag is specified.

R=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 22:17:43 +00:00
aluebs@webrtc.org
0b72f5863b Add experimental noise suppression dummy API. 
Add this flag to the voe_cmd_test.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-19 15:17:51 +00:00
minyue@webrtc.org
cc92e000f3 1. adding request of ACM version in the manual mode of voe_auto_test 
2. adding command line flag for automated mode of voe_auto_test to choose between ACMs

3. adding request of ACM version in voe_cmd_test

R=phoglund@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2281004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-30 08:43:50 +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
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
andrew@webrtc.org
ea83c6ac9d Allow voe_cmd_test to select Opus mono (now the default). 
* Opus handles stereo and mono on the same payload type, so we need a different mechanism to choose between them.
* Assorted cleanups.

BUG=webrtc:1710
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 15:57:36 +00:00
pwestin@webrtc.org
835dbf4516 Fix no received audio in tests. 
BUG=1582, 1581
Review URL: https://webrtc-codereview.appspot.com/1281005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 17:24:15 +00:00
pwestin@webrtc.org
e30823911c Move the VoE tests to use external transport instead of the built in udp transport 
Review URL: https://webrtc-codereview.appspot.com/1223006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 16:12:57 +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
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
andrew@webrtc.org
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
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