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Author SHA1 Message Date
andrew@webrtc.org
07ebdb9432 Handle 96 kHz when downmixing the capture path. 
BUG=issue721
TEST=96 kHz capture on Windows works.

Review URL: https://webrtc-codereview.appspot.com/722004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-03 18:03:02 +00:00
mflodman@webrtc.org
10a31520a5 Disabled FileBeforeStreamingTest.TestStartPlayingFileLocallyWithStartPlayout. 
BUG=719

TBR=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/710007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-03 09:50:15 +00:00
wu@webrtc.org
792e974949 Refactor the public interfaces to use the full path in include. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/708006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2546 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-01 22:14:51 +00:00
andrew@webrtc.org
d7a71d0719 Prepare to roll Chromium to 149181. 
- This roll brings in VS2010 by default. The buildbots
  need updating (issue710).
- We'll roll to 149181 later (past current Canary) to fix
  a Mac gyp issue:
  https://chromiumcodereview.appspot.com/10824105
- Chromium is now using a later libvpx than us. We should
  investigate rolling our standalone build.
- Fix set-but-unused-warning
- Fix -Wunused-private-field warnings on Mac.

TBR=kjellander@webrtc.org
BUG=issue709,issue710
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/709007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-01 01:40:02 +00:00
andrew@webrtc.org
6f8db36e04 Reorganize voice_engine/. 
The usual changes:
voice_engine/main/source -> voice_engine/
voice_engine/main/interface -> voice_engine/include
voice_engine/main/test -> voice_engine/test
Include path changes.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/705004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-27 21:49:28 +00:00
tommi@webrtc.org
a9da4c55ef Landing for thakis. Original review here: 
https://webrtc-codereview.appspot.com/667013/
Review URL: https://webrtc-codereview.appspot.com/701004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-20 11:17:23 +00:00
stefan@webrtc.org
ddfdfed3b5 Pass capture time (wallclock) to the RTP sender to compute transmission offset 
- Change how the transmission offset is calculated, to
  incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
  We must use the same clock as in the RTP module to be able to measure
  the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/666006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 13:21:22 +00:00
andrew@webrtc.org
4ecea3e105 Downmix before resampling in capture and render paths. 
We previously had an error when a mono capture device was used with
a stereo codec. This is prevented by avoiding any remixing in
AudioProcessing. Instead, capture side downmixing is now done before
resampling. Upmixing can now be handled properly by AudioCoding,
since the AudioProcessing error condition has been removed.

On the render side, downmixing now occurs before resampling. Ideally
this would be handled still earlier in the chain. Similarly, downmixing
for the AudioProcessing reference data occurs before resampling. This
code has been refactored into RemixAndResample, with a comprehensive
unittest added in output_mixer_unittest.cc.

BUG=issue624
TEST=manually through voe_cmd_test, by using mono and stereo capture
and render devices with mono and stereo codecs. voice_engine_unittest,
voe_auto_test.

Review URL: https://webrtc-codereview.appspot.com/676004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 03:25:31 +00:00
andrew@webrtc.org
81cf5e4752 Move test to src/test. 
- Refer to top-level directories by <(DEPTH), e.g. <(DEPTH)/testing.
- Remove now unneeded third_party_root.

TBR=henrike@webrtc.org
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/669007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 01:41:54 +00:00
henrike@webrtc.org
643be71700 Adds variable for third party directory. 
BUG=348
TEST=Manual testing in Chrome and WebRTC workspace.

Review URL: https://webrtc-codereview.appspot.com/674005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-25 10:48:58 +00:00
tnakamura@webrtc.org
b9c1833c2c Add channel info to the Actions->Codec Changes menu in the VoE test app. 
Review URL: https://webrtc-codereview.appspot.com/665005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-22 16:29:38 +00:00
braveyao@webrtc.org
77e18124f9 Fix the flakiness in FileBeforeStreamingTest 
BUG = 619
TEST = voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/658006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-22 10:41:11 +00:00
kjellander@webrtc.org
5608fe9861 Disabling FileBeforeStreamingTest due to flakiness. 
BUG=619
TBR=xians1
TEST=Tested on Linux, Mac and Windows.

Review URL: https://webrtc-codereview.appspot.com/654006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-20 06:14:31 +00:00
braveyao@webrtc.org
dfa6b697e2 Refine the error handling made in rev2373 
Review URL: https://webrtc-codereview.appspot.com/644005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-19 06:38:59 +00:00
henrika@webrtc.org
37198007ea GetRecPayloadType now logs a warning instead of and error when the user asks for the payload type while no packets have been received. 
BUG=605
TEST=

Review URL: https://webrtc-codereview.appspot.com/660004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 11:00:12 +00:00
braveyao@webrtc.org
4de777ba2b Refine the error processing of StopRecordingMicrophone. 
BUG = 
TEST = 
Review URL: https://webrtc-codereview.appspot.com/636007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-15 02:37:53 +00:00
turaj@webrtc.org
bdf7ee5bab This simple change should adress issue 471. 
Previously I uploaded patch 640007 to address issue 471. Today, while discussing that patch with Andrew, we noticed this patch should do the job. Leo is not here to verify it, but Andrew did some test to verify it. I'll ask Leo to do some testing. 

We don't want to abandon patch 640007 as it will save some complexity. 
Review URL: https://webrtc-codereview.appspot.com/648004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-14 23:46:35 +00:00
braveyao@webrtc.org
b0bcf13dd4 Trival fix to relative paths of audio files in voe_ui_win_test 
BUG  = 
TEST = voe_ui_win_test
Review URL: https://webrtc-codereview.appspot.com/635005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 02:21:44 +00:00
braveyao@webrtc.org
ab12990b1b In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately. 
Now we would get a failure if we do the same thing and the file would not be played out. Then GTalk/Hangout also reported this failure to us. 
This CL is to restore the original function. 

BUG = Issue 490
TEST = Manual test and voe_auto_test->FileBeforeStreamingTest
Review URL: https://webrtc-codereview.appspot.com/569016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-04 03:26:39 +00:00
tina.legrand@webrtc.org
4517585db5 Adding separate payload types for stereo modes 
BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 09:27:35 +00:00
andrew@webrtc.org
16fcb247b2 Disable flaky VolumeTests only on Linux. 
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/611005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-30 17:26:32 +00:00
andrew@webrtc.org
459955f821 Move audio_frame_operations to the utility module. 
TBR=henrika@webrtc.org
BUG=issue551
TEST=voe_auto_test, webrtc_utility_unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/599006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 22:13:14 +00:00
andrew@webrtc.org
aafa49bb85 Disable flaky VolumeTest.DefaultSpeakerVolumeIsAtMost255. 
This test failed on six CLs in a row recently.

TBR=xians@webrtc.org
BUG=issue367
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/595007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 22:05:15 +00:00
phoglund@webrtc.org
dbaa893525 Completed rewrite of APM extended test. 
Removed NS tests since they are already covered by audio_processing_test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/603004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-28 14:36:59 +00:00
leozwang@webrtc.org
351fb6d3b4 Exclude code that don't work on android in voe_cmd_test 
Description:
Ths cl makes voe_cmd_test work on android by excluding some code
that are availabel on android today, some highlights
1. change maxnumofchannles
2. disable audio device selection
3. disable set/get volume

BUG=
TEST=test on try bots
Review URL: https://webrtc-codereview.appspot.com/584009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-25 16:47:35 +00:00
andrew@webrtc.org
f45d47ad7d Remove mixing tests from voe_extended_test.cc 
These have been moved to:
src/voice_engine/main/test/auto_test/standard/mixing_test.cc

BUG=
TEST=build voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 17:27:59 +00:00
andrew@webrtc.org
51b4f3e6a8 Try to fix MixingTest on the Win bots. 
- Relax the constraints on recording duration.
- Remove unneeded file deletes. (These files will be properly
  overwritten anyway).

TBR=henrike@webrtc.org
BUG=issue534
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/600006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 17:26:05 +00:00
mflodman@webrtc.org
6af9594d71 Added gyp variable to include/exclude all tests. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/597004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 13:23:35 +00:00
niklas.enbom@webrtc.org
ee646c37d4 I know this is ugly, but it helps a lot to quickly update webRTC in Chrome and libJingle. 
Review URL: https://webrtc-codereview.appspot.com/596004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-24 11:41:02 +00:00
andrew@webrtc.org
7fbfc4ce79 Use correct variable in trace. 
TBR=leozwang@webrtc.org
TEST=build

Review URL: https://webrtc-codereview.appspot.com/593004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 22:22:36 +00:00
andrew@webrtc.org
9dc45dad1b Move trunk/test/data -> trunk/data 
BUG=
TEST=all trybot test failures passed locally

Review URL: https://webrtc-codereview.appspot.com/583007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 15:39:01 +00:00
andrew@webrtc.org
a1a34d675f Avoid flakiness by skipping more output verification. 
- Add a SCOPED_TRACE in case it flakes out again.
- The test doesn't need to be very long, so shorten it to save the bots some time.

TBR=henrike@webrtc.org
BUG=
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/588006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-23 00:45:00 +00:00
andrew@webrtc.org
294be77c2e Permit mixing mono and stereo streams. 
Add mixing tests based on older ones from the extended tests.

BUG=issue534
TEST=manual, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/576014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 03:28:41 +00:00
phoglund@webrtc.org
1ad477de3e Added a bit flip fuzz test to the voice engine. 
Extracted encryption classes to a new test library.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/564009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-18 08:02:37 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module. 
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 11:08:54 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 23:01:04 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 12:33:50 +00:00
andrew@webrtc.org
270e9db039 Clarify the impact of disabling VAD on DTX. 
TBR=henrika@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/566009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 19:09:03 +00:00
niklas.enbom@webrtc.org
f6edfeff63 Adding one parameter to typing detection tuning 
Review URL: https://webrtc-codereview.appspot.com/569009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-09 13:16:12 +00:00
andrew@webrtc.org
e59a0aca6a Fix AudioFrame types. 
volume_ is not set anywhere so I'm removing it.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-08 17:12:40 +00:00
andrew@webrtc.org
589673f1cb Fix volume setting while not playing on PulseAudio. 
We now only set the volume when starting playout if the user has called
SetSpeakerVolume while we weren't playing. This now also ensures it will
actually get set to what the user requested rather than being overridden
by a default value.

Add tests to voe_auto_test.

BUG=6140661
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/566006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 21:42:49 +00:00
braveyao@webrtc.org
ba0f9fe10b Trival fix to voe_auto_test according to the main source codes 
BUG = NULL
TEST = voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/554004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-07 10:06:43 +00:00
andrew@webrtc.org
63a509858d Rename AudioFrame members. 
BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/542005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 23:56:37 +00:00
phoglund@webrtc.org
719dba7e79 Further cleaned up voe_standard_test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/522003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 07:32:37 +00:00
andrew@webrtc.org
a88cb6fce0 Add HighPassFilter and StereoChannelSwapping tests. 
BUG=issue451
TEST=voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/531001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 17:00:20 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
andrew@webrtc.org
9c4f6a5ff9 Add an AudioFrameOperations unittest. 
Additionally, reformat audio_frame_operations to Goog style.

BUG=issue451
TEST=voice_engine_unittests

Review URL: https://webrtc-codereview.appspot.com/528001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 22:32:03 +00:00
tommi@webrtc.org
a990e122c4 * Change the reference counting implementation for VoE to be per object and 
not per interface. This simplifies things a bit, reduces code and makes it
  possible to implement reference counting (if we ever want) without the
  static Delete() method.  (Reference counted objects are traditionally
  implicitly deleted via the last Release())

* Since the reference counting code is now simpler, there's no need for the
  RefCount class so I'm removing it.

* Also removing the optional |ignoreRefCounters| variable from the VoiceEngine::Delete
  method.  The justification is that it's no longer used and the reason it was there
  in the first place was to avoid bugs in third party code, so it's an indication that
  something is wrong elsewhere.

* Fix a crash in voe_network_impl and voe_extended_test that would happen on machines with IPv6 support.

* Fix an assert (handle the case) in the extended audio tests when SetCNAME is called with a NULL pointer.

* As a side effect of this change, hopefully the footprint of VoE will be slightly smaller :)

BUG=10445 (Coverity)
Review URL: https://webrtc-codereview.appspot.com/521003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 15:28:22 +00:00
andrew@webrtc.org
1c7bfe02f7 Fail silently when swapping mono. 
TBR=tina.legrand@webrtc.org
BUG=issue451
TEST=forthcoming unittest

Review URL: https://webrtc-codereview.appspot.com/527003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 00:20:28 +00:00
andrew@webrtc.org
02d7174722 Add API to swap stereo channels. 
BUG=issue451
TEST=manually with voe_cmd_test, using stereo and mono codecs

Review URL: https://webrtc-codereview.appspot.com/519001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 19:47:00 +00:00