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79 Commits

Author SHA1 Message Date
andrew@webrtc.org
50b2efef6e Add a wrapper around PushSincResampler and the old Resampler. 
The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 17:27:29 +00:00
henrika@webrtc.org
4392d5f9f8 Fix for "RTP dynamic payload type 100 is reserved" 
TBR=perkj
BUG=227036 (in crbug.com)

TEST=out\Debug\voe_auto_test.exe --automated --gtest_filter=Dtmf* where I
manually modified the test and used 100 as new PT (which I first verified was
already used by CN, 48000).

BUG=

Review URL: https://webrtc-codereview.appspot.com/1319010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 07:34:25 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files 
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 12:40:34 +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
6faf71d27b Remove the old unused udp_transport 
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 23:25:25 +00:00
henrika@webrtc.org
19da719a5f Resolves TSan v2 reports data races in voe_auto_test. 
--- Note that I will add more fixes to this CL ---

BUG=1590

Review URL: https://webrtc-codereview.appspot.com/1286005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 14:34:57 +00:00
pwestin@webrtc.org
b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use 
Review URL: https://webrtc-codereview.appspot.com/1203009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:51:42 +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
henrika@webrtc.org
aa527bbc91 Disabling MixingTests due to race conditions. 
BUG=1580
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1285005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 15:19:10 +00:00
henrika@webrtc.org
bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC 
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:39:09 +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
henrika@webrtc.org
0746ce1465 Fixes memory leak in AudioLevel class reported by memory try bots. 
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1275008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 11:58:12 +00:00
henrika@webrtc.org
d108a46206 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer 
BUG=225690

Review URL: https://webrtc-codereview.appspot.com/1269008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 11:25:31 +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
1b31c78e5f Remove VoE's default call in Trace::SetLevelFilter. 
This is an application level setting. Applying it here has the potential to override the application's preferences.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:09:48 +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
sjlee@webrtc.org
b4c441a785 Enable the below APIs for iOS. 
class VoEAudioProcessing
  int RegisterRxVadObserver();
  int DeRegisterRxVadObserver();
  int SetEcMetricsStatus();
  int GetEcMetricsStatus()
  int GetEchoMetrics();
  int GetEcDelayMetrics();

class VoENetEqStats
  int GetNetworkStatistics();

class VoEVolumeControl
  int SetChannelOutputVolumeScaling();
  int GetChannelOutputVolumeScaling();
Review URL: https://webrtc-codereview.appspot.com/1159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 11:12:20 +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
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
andrew@webrtc.org
a9a1df0035 Remove the error return on SetAGC failure introduced by r3605. 
BUG=webrtc:1464

Review URL: https://webrtc-codereview.appspot.com/1166005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 23:36:10 +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
andrew@webrtc.org
2412085bc1 Don't upsample the capture signal early. 
* Remove the unneeded _mixingFrequency.
* Rename CheckForSendCodecChanges to better elucidate its function.
* Remove an unnecessary memcpy.

Upsampling should be done late in the chain. This is practically relevant
on mobile, where the capture rate is fixed at 16 kHz. When using Opus, the
signal was upsampled to 32 kHz and was no longer compatible with AECM, which only supports up to 16 kHz.

NEEDS_QA=true
TEST=run calls with a variety of capture device rates and codecs
BUG=chromium:178040,webrtc:1446

Review URL: https://webrtc-codereview.appspot.com/1146004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-02 00:14:46 +00:00
andrew@webrtc.org
6be1e934ad Properly error check calls to AudioProcessing. 
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.

BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 18:47:28 +00:00
leozwang@webrtc.org
9ee5a4ccd8 Enable External MediaProcessing on Mobile 
Bug=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1133005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 17:19:35 +00:00
tina.legrand@webrtc.org
7a7a008031 Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Committed: https://code.google.com/p/webrtc/source/detail?r=3543

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 10:27:48 +00:00
fischman@webrtc.org
f61e02c81f Misc cleanups to webrtc/android code: 
- Replace some deprecated calls/enums with their more modern equivalents.
- Clean up some usage of global data and/or hide it better
- Catch specific exceptions instead of Exception, and log the exception instead
  of just its message.
- Random log message cleanups
- Added a build_with_libjingle gyp variable to mimic build_with_chromium for
  when webrtc is built as part of a libjingle project but not part of chromium.

BUG=webrtc:1169
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1105010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 23:13:46 +00:00
tina.legrand@webrtc.org
eb7ebf20ed Revert 3543 
> Changing non-const reference arguments to pointers, ACM
> 
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
> 
> BUG=issue1372
> 
> Review URL: https://webrtc-codereview.appspot.com/1103012

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:57:31 +00:00
tina.legrand@webrtc.org
374aa49e1a Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:22:23 +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
andrew@webrtc.org
1e7ed7afe9 Use LOG_F interface for unsupported functions. 
This will provide the function name in the log.

BUG=b/8115521
TESTED=enabled ANDROID_NOT_SUPPORTED on Linux and observed log lines as expected

Review URL: https://webrtc-codereview.appspot.com/1096005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-05 21:23:39 +00:00
kjellander@webrtc.org
fa53d8717c Fixing/disabling Windows x64 warnings 
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.

With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.

BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64

Review URL: https://webrtc-codereview.appspot.com/1060008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 10:07:17 +00:00
andrew@webrtc.org
6ed8ebcef9 Fix MaxChannels test; 32 -> 100. 
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/1060010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-02 00:05:58 +00:00
andrew@webrtc.org
4a6f62d4dc Remove (in practice) the voice engine channel limit. 
There's really no reason for this limit. I've bumped it to a
practically unreachable ceiling, with a TODO for removing it
entirely.

TBR=henrika
BUG=b/8122300

Review URL: https://webrtc-codereview.appspot.com/1070014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 23:42:44 +00:00
turaj@webrtc.org
9d532fd275 Fix propagating RED paylaod-type to ACM. 
BUG=issue1322
TBR=henrika@google.com
Review URL: https://webrtc-codereview.appspot.com/1086005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 18:34:19 +00:00
turaj@webrtc.org
8c8ad85c5d fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). 
issue=1322 
test=trybot, voe auto-tes
Review URL: https://webrtc-codereview.appspot.com/1043007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 18:20:17 +00:00
henrike@webrtc.org
09738616de Fixes payload spelling error. 
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1052006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 16:43:45 +00:00
andrew@webrtc.org
ae1a58bba4 Replace AudioFrame's operator= with CopyFrom(). 
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.

Review URL: https://webrtc-codereview.appspot.com/1031007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 04:44:30 +00:00
wjia@webrtc.org
a3c82bf667 Remove '<(library)' in gyp files. 
This will remove all usage of '<(library)' in all webrtc gyp files. 
Review URL: https://webrtc-codereview.appspot.com/1049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 23:42:21 +00:00
niklas.enbom@webrtc.org
218c542c0b Make VoE handle longer delays 
Review URL: https://webrtc-codereview.appspot.com/1047004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 22:25:49 +00:00
wjia@webrtc.org
3f9db3735e Fix android clang build. 
Android clang build complains about unused private field.
Review URL: https://webrtc-codereview.appspot.com/1025006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-12 01:09:03 +00:00
andrew@webrtc.org
a5e7e76def Use %d for signed value in trace. 
BUG=1259
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/1028007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 17:11:50 +00:00