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Author SHA1 Message Date
leozwang@webrtc.org
2a84f63719 Rename android file name 
Rename file name to follow code style.

BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/867004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-03 21:40:06 +00:00
leozwang@webrtc.org
e4ba864368 Fix building error and rename java class name 
1. Fix building error because of r2804
2. Rename java class name to WebRTCAudioDevice, so it's more meaningful
to 3rd party devleoper

BUG=None
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/821006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-24 18:46:05 +00:00
andrew@webrtc.org
236d5d3159 Reorganize audio_device to the standard layout. 
Review URL: https://webrtc-codereview.appspot.com/831004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-21 20:46:40 +00:00
leozwang@webrtc.org
cf1375a1f1 Make SetAndroidAudioDeviceObjects return 0 
Description:
Make SetAndroidAudioDeviceObjects return 0 so application can work with both java
and opensl implementation without code change.

BUG=None
TEST=trybot
Review URL: https://webrtc-codereview.appspot.com/817004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-21 17:39:45 +00:00
leozwang@webrtc.org
81cd447219 Enable SetRecordDevice on Android 
This api is very critical to make aec work properly, although
it's only available in audio device java implementation, will
add to opensl es in future.

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/820004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-20 17:18:51 +00:00
leozwang@webrtc.org
2db85bcba7 Make webrtc build with audio device java impl and add an option to enable it 
BUG=
TEST=buildbots

This cl is to make audio device java implemenation build in webrtc, and add an
option in gyp so we can switch between opensl implementaiton and java
implementation.
Review URL: https://webrtc-codereview.appspot.com/801004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-18 20:19:00 +00:00
andrew@webrtc.org
0be1f234b6 Add merge_libs_dependencies and remove voice_engine_dependencies. 
TBR=wu,turaj
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/798006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-15 02:50:52 +00:00
sjlee@webrtc.org
414fa7f0c4 Change MAC_IPHONE to WEBRTC_IOS. 
Review URL: https://webrtc-codereview.appspot.com/788004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 17:25:46 +00:00
sjlee@webrtc.org
4b42508cc0 This CL is WebRTC VoiceEngine for iOS and is from CL713004. 
After patching this, first comments some video related lines in webrtc.gyp and src/module/module.gyp
And then do the below command.

$> ./build/gyp_chromium --depth=.  -DOS=ios -Dtarget_arch=armv7 -Dinclude_tests=0 -Denable_protobuf=0 -Denable_video=0 webrtc.gyp
$> xcodebuild -sdk iphoneos [-configuration Release]
Review URL: https://webrtc-codereview.appspot.com/768009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 17:58:21 +00:00
kma@webrtc.org
0221b78e2e Added run time ARM-Neon detection feature in SPL functions. 
Review URL: https://webrtc-codereview.appspot.com/728010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-08 00:09:26 +00:00
andrew@webrtc.org
7692239b18 Work around bot filesystem flakiness in MixingTest. 
TBR=braveyao

Review URL: https://webrtc-codereview.appspot.com/780004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-07 05:18:48 +00:00
andrew@webrtc.org
cc53b7c97b Disable test causing race conditions. 
TBR=kjellander
BUG=issue788

Review URL: https://webrtc-codereview.appspot.com/770004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 06:55:21 +00:00
andrew@webrtc.org
b93522857c Trivial fix for memcheck error. 
TBR=xians

Review URL: https://webrtc-codereview.appspot.com/763005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 04:23:14 +00:00
andrew@webrtc.org
55c0d4a683 Add support for clock drift compensation. 
Support clock drift compensation on Windows and add an API to allow
enabling dynamically.

BUG=issue773
TEST=unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/744007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 02:13:12 +00:00
henrika@webrtc.org
8a2fc88459 Adds new GetRemoteRTCPSenderInfo() and GetRemoteRTCPReportBlocks APIs to VoE. 
BUG=559
TEST=manual tests using Windows UI client.

Review URL: https://webrtc-codereview.appspot.com/735011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 08:53:55 +00:00
vikasmarwaha@webrtc.org
bdb03d48ae Fix for issue 420 in TransmitMixer::SetTypingDetectionParameters. 
Review URL: https://webrtc-codereview.appspot.com/747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 17:46:09 +00:00
andrew@webrtc.org
9ea1be81d8 Remove unnecessary failure on changing CN payload type while sending. 
BUG=issue625

Review URL: https://webrtc-codereview.appspot.com/731009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2630 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-17 15:49:07 +00:00
andrew@webrtc.org
cb53410877 Make some dependencies more flexible. 
BUG=none
TEST=trybot

Review URL: https://webrtc-codereview.appspot.com/728005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-09 18:28:40 +00:00
braveyao@webrtc.org
743e5cf6b7 remove flaky test case in FileBeforeStreamingTest 
BUG = Issue 719
TEST = VoE standard test
Review URL: https://webrtc-codereview.appspot.com/718006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-08 06:07:39 +00:00
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