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Author SHA1 Message Date
pbos@webrtc.org
5b10d8fb18 Fix some voe_auto_test uninitialised-value errors. 
BUG=
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 15:50:07 +00:00
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. 
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 13:39:27 +00:00
stefan@webrtc.org
9de89a6f6b Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. 
R=pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 12:42:15 +00:00
pbos@webrtc.org
08933a5dfb Initialize payload-type frequency in channel.cc. 
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.

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

Review URL: https://webrtc-codereview.appspot.com/1780004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 10:06:29 +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
andrew@webrtc.org
0851df8d60 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. 
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.

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

Review URL: https://webrtc-codereview.appspot.com/1683004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 17:03:47 +00:00
kjellander@webrtc.org
6c35e0b0f7 Reorganize test targets in WebRTC 
This CL will lower the number of test targets in WebRTC by:

Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests

Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests

Merge into test_support_unittests:
* channel_transport_unittests

channel_transport.gyp was also removed in favor for test.gyp.

I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.

Buildbot configuration update will be synced with the commit of this CL.

TEST=trybots
BUG=1843
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-11 08:29:17 +00:00
pwestin@webrtc.org
1064cf06b0 Fixed Rtp/Rtcp tests 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1627005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 16:03:19 +00:00
andrew@webrtc.org
da710448b2 Fix size_t to int conversion error on Win64. 
TBR=pwestin

Review URL: https://webrtc-codereview.appspot.com/1626005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 01:43:12 +00:00
pwestin@webrtc.org
8d80fa83fc Fix for STL vector function data not available. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1626004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 21:33:06 +00:00
pwestin@webrtc.org
d30859e58e Connect ACM with RTP module for audio NACK. 
Depends on http://review.webrtc.org/1507004/

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1613007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 21:09:01 +00:00
pwestin@webrtc.org
db24995680 Wire up Nack for Voe 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1614004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 15:33:20 +00:00
andrew@webrtc.org
0a38432ea5 Fix error in mixing test for supported sample rates. 
With the switch to an arbitrary resampler, we now support these strange
rates.

TBR=turaj

Review URL: https://webrtc-codereview.appspot.com/1604004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 22:52:09 +00:00
wu@webrtc.org
fa64a595ad Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. 
This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.

BUG=1828
TEST=unit tests
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1598005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 21:27:57 +00:00
andrew@webrtc.org
c1eb560a5c Replace the old resampler with SincResampler in the voice engine signal path. 
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.

BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1590004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 19:00:29 +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
pbos@webrtc.org
9aca5b34e1 Remove #pragma once 
BUG=1830
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:19:09 +00:00
stefan@webrtc.org
a5cb98cbbd Breaking out RTP header parsing from the RTP module. 
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 12:12:51 +00:00
andrew@webrtc.org
f791b1cebf Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1574004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 00:38:02 +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
niklas.enbom@webrtc.org
3be565b502 Refactoring for typing detection 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 21:04:24 +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
andrew@webrtc.org
8c845cb623 Relax VoE's max packet length threshold. 
The earlier threshold would cause packets from a currently available
codec (L16, 32 kHz, stereo) to be discarded.

TESTED=voe_cmd_test using L16, 32 kHz, stereo now works properly.
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1305008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 15:28:02 +00:00
phoglund@webrtc.org
258f55efc0 Disabled flaky test. 
BUG=1719
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 12:35:00 +00:00
andrew@webrtc.org
28e82bfec6 Replace Resampler with PushResampler in transmit_mixer. 
* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.

BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 00:30:36 +00:00
andrew@webrtc.org
342353780d Consolidate common_audio into a single target. 
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.

R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:43:26 +00:00
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