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26 Commits

Author SHA1 Message Date
braveyao@webrtc.org
83a062cc5f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() 
BUG=1891
Test=ManualTest

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

Review URL: https://webrtc-codereview.appspot.com/1622004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 08:09:05 +00:00
andrew@webrtc.org
569fdef732 Revert some variables to uint32_t to fix compile errors on Mac gcc. 
TBR=xians

Review URL: https://webrtc-codereview.appspot.com/1633004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-08 00:43:25 +00:00
andrew@webrtc.org
6f69eb78dd Allow audio devices with up to 64 channels on Mac. 
Does not increase memory requirements. Adds an additional check to ensure
configurations requiring more memory per IO block than the input ring buffer
contains are rejected.

BUG=1904
TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality.
Selecting a higher sample rate (96 kHz), which would otherwise give choppy
audio, instead results in an error.

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

Review URL: https://webrtc-codereview.appspot.com/1628004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4198 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 17:56:50 +00:00
fischman@webrtc.org
34a77354a8 Removed unused class members to enable clang=1 android build. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1275
TESTED=video_demo_apk builds with clang=1
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1605004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 00:37:21 +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
braveyao@webrtc.org
c93b1d038d CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin 
BUG=
TEST=voe_auto_test
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 10:14:56 +00:00
niklas.enbom@webrtc.org
e2a800644c Linux support for typing detection 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 21:33:11 +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
xians@webrtc.org
233c58de47 Landing 1399004, Minor clean up on the un-used _measureDelay code 
Those code is/will never used, removing it makes the code better.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@3961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 11:52:47 +00:00
andrew@webrtc.org
a31c428307 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. 
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.

Our ALSA interface always requires 48 kHz, allowing ALSA to handle resampling.

This also removes WEBRTC_PA_GTALK which was not defined anywhere.

BUG=webrtc:1395
TESTED=Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good. Testing AEC was difficult as I can't find a way to change the sample rate of an individual device in PulseAudio. Using a webcam at 32 kHz, other problems were the overriding contribution to quality degradation (delay issues, possible clock drift from the camera). At least I verified that the quality got no worse with this patch.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 19:01:46 +00:00
andrew@webrtc.org
7cb766b016 Remove 44.1 kHz workaround from AudioDevice on WASAPI. 
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.

BUG=webrtc:1395
TESTED=Set capture device to 44.1 and render device to 48 and vice versa and observed good AEC. The quality is considerably worse before this change. Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1383004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 18:56:38 +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
braveyao@webrtc.org
c14b728b71 Add lock to prevent possible rare race condition in Win coreAudio capture implementation. 
BUG = 
TEST = voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1320011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 09:32:07 +00:00
pbos@webrtc.org
ab9202b673 Removing remaining WebRtc_Word32 not in typedefs.h 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1306006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:59:17 +00:00
pbos@webrtc.org
2550988baa WebRtc_Word32 -> int32_t in audio_device/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:30:35 +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
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
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
andrew@webrtc.org
73a702c979 This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware. 
Review URL: https://webrtc-codereview.appspot.com/1061007
Patch from Gil Osher <gil.osher@vonage.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-30 21:18:31 +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
leozwang@webrtc.org
515ef2428c Clean up variable after it gets deleted 
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/939038

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 05:36:36 +00:00
braveyao@webrtc.org
72feb0b2e2 Not to enum NOTPRESENT audio devices with CoreAudio on Win 
BUG = 
TEST = Manual test
Review URL: https://webrtc-codereview.appspot.com/939037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 22:36:07 +00:00
henrika@webrtc.org
5ba3decc94 Ensures that we can build using VS 2012 on Windows. 
See more details at https://code.google.com/p/webrtc/issues/detail?id=1146&

TBR=Niklas
BUG=1146

Review URL: https://webrtc-codereview.appspot.com/939028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 09:12:02 +00:00
leozwang@webrtc.org
06d72d881f Add Android OWNER files 
Message:
Add OWNER files so I can review and approve changes for Android.
I also should be owner for all .mk file, but it's OK for now,
please review.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/932016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 17:51:55 +00:00
leozwang@webrtc.org
5f9970fb0b Refactor OpenSL audio driver 
Message:
I want to start this review, the basica framework is almost done.

Description:
This implementation is similar to current one, but
1. followed the design doc at
https://docs.google.com/a/google.com/document/d/1g5q2SVtkFPl2OSjvSF3eeLb_S7sCnbINxiBUES6XLCM/edit
which uses two threads, playout thread and recording thread, uses large
audio buffer, etc.
2. google code style.

What are missing in this cl,
1. a better way to control schedule/thread priority.
2. java/jni interface to better support what cannot be done in OpenSL.

Please take a review, thanks.
Review URL: https://webrtc-codereview.appspot.com/902004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-02 16:48:03 +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