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Author SHA1 Message Date
henrika
8324b525dc Adding playout volume control to WebRtcAudioTrack.java. 
Also adds a framework for an AudioManager to be used by both sides (playout and recording).
This initial implementation only does very simple tasks like setting up the correct audio
mode (needed for correct volume behavior). Note that this CL is mainly about modifying
the volume. The added AudioManager is only a place holder for future work. I could have
done the same parts in the WebRtcAudioTrack class but feel that it is better to move these
parts to an AudioManager already at this stage.

The AudioManager supports Init() where actual audio changes are done (set audio mode etc.)
but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the
case here. Hence, the AM now serves as the center for getting audio parameters and then inject
these into playout and recording sides. Previously, both sides acquired their own parameters
and that is more error prone.

BUG=NONE
TEST=AudioDeviceTest
R=perkj@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45829004

Cr-Commit-Position: refs/heads/master@{#8875}
 2015-03-27 09:56:35 +00:00
tommi@webrtc.org
38492c5b6f Re-land 8810 "- Add a SetPriority method to ThreadWr..." 
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
> 
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> > 
> > BUG=
> > R=magjed@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44729004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/48609004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459005

Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:42:46 +00:00
tommi@webrtc.org
90a1cb4630 Revert 8810 "- Add a SetPriority method to ThreadWrapper" 
Seeing if this is causing roll issues.

> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
> 
> BUG=
> R=magjed@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44729004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48609004

Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:34:46 +00:00
andrew@webrtc.org
3200a64b3c Minor fix for MIPS Android build. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

Cr-Commit-Position: refs/heads/master@{#8813}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 22:55:43 +00:00
tommi@webrtc.org
b6817d793f - Add a SetPriority method to ThreadWrapper 
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional

BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44729004

Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 15:52:43 +00:00
henrika@webrtc.org
80d9aeeda5 Adds full-duplex unit test to AudioDeviceTest on Android 
BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42709004

Cr-Commit-Position: refs/heads/master@{#8795}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 15:28:42 +00:00
tommi@webrtc.org
361981faa8 Use scoped_ptr for ThreadWrapper::CreateThread. 
BUG=
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45799004

Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:45:42 +00:00
pbos@webrtc.org
86639737b8 Remove thread id from ThreadWrapper::Start(). 
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.

BUG=4413
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43699004

Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 00:07:45 +00:00
henrika@webrtc.org
74d4792af5 Fixes issue in RunPlayoutWithFileAsSource related to uninitialized member 
BUG=4408
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45609004

Cr-Commit-Position: refs/heads/master@{#8668}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 11:59:19 +00:00
henrika@webrtc.org
474d1eb223 Adds C++/JNI/Java unit test for audio device module on Android. 
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.

It also:

- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.

BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40069004

Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-09 12:40:43 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
henrika@webrtc.org
0a3ff7976b New AudioTrack implementation now works on pre-Lollipop devices. 
The previous version used an AudioTrack.write() implementation that required API Level 21. This is now fixed.

BUG=4339
R=magjed@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42459004

Cr-Commit-Position: refs/heads/master@{#8494}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-25 09:28:20 +00:00
henrika@webrtc.org
962c62475e Refactoring WebRTC Java/JNI audio track in C++ and Java. 
This CL is part II in a major refactoring effort. See https://webrtc-codereview.appspot.com/33969004 for part I.

- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioTrack (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Simplified the delay estimate
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup

Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).

BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39169004

Cr-Commit-Position: refs/heads/master@{#8460}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 11:54:41 +00:00
pbos@webrtc.org
d5ce2e63df Remove EventWrapper::Reset(). 
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

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

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:58:38 +00:00
henrika@webrtc.org
58f6f01acc WebRTC now compiles for enable_android_opensl=1. 
Default is enable_android_opensl=0 but we should build for OpenSL as well.

BUG=4293
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40719004

Cr-Commit-Position: refs/heads/master@{#8360}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 11:36:12 +00:00
perkj@webrtc.org
a9eaeebc6a Fix problem where Android VoE can not record on multiple channels. 
The issue was introduced in https://webrtc-codereview.appspot.com/33969004/
R8325

TEST=  Build libjingle_peerconnection_android_unittest and then run "CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest"
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38089004

Cr-Commit-Position: refs/heads/master@{#8349}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 12:33:28 +00:00
henrika@webrtc.org
62f6e75673 Refactoring WebRTC Java/JNI audio recording in C++ and Java. 
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:

- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup

Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).

BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33969004

Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 08:39:19 +00:00
tommi@webrtc.org
875c97ed9d Remove SetNotAlive method from the thread class. 
Also cleaning up methods with the same name in other classes that are derived from the above method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41759004

Cr-Commit-Position: refs/heads/master@{#8242}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-04 11:12:39 +00:00
tommi@webrtc.org
4161715e3f Remove ChangeUniqueID. 
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.

It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.

BUG=
R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37849004

Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 12:14:13 +00:00
henrika@webrtc.org
a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer 
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 16:22:09 +00:00
henrika@webrtc.org
dd43bbed8f Volume buttons in AppRTCDemo should affect output audio volume (part II). 
See https://webrtc-codereview.appspot.com/32399004/ for part I.

BUG=3279
TEST=AppRTC demo
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 15:48:05 +00:00
kjellander@webrtc.org
b96ea2aab5 Remove former team members from OWNERS and WATCHLISTS 
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@

BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 06:12:08 +00:00
solenberg@webrtc.org
c6db88b0cf Make it possible to build webrtc for arm64. 
- Bump revision of protobuf lib
- Remove -Wextra for arm64 gcc targets (warnings in stlport)
- Add MemoryBarrier implementation in single_rw_fifo.cc.
- [pending 15619004]: Bump revision of /deps/tools/android to get md5sum_bin for arm64.

BUG=chromium:354405,chromium:354539
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 17:15:42 +00:00
henrike@webrtc.org
14abcc7322 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. 
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 16:54:44 +00:00
andrew@webrtc.org
8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>. 
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:10:28 +00:00
andrew@webrtc.org
c7c432aa9b Remove AudioDevice::{Microphone,Speaker}IsAvailable. 
This was only used for logging, except on Mac, where the methods are
now private.

BUG=3132
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 16:49:26 +00:00
henrike@webrtc.org
32c26eb90b Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback 
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-17 23:12:51 +00:00
henrike@webrtc.org
ead202b973 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. 
BUG=2801
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 23:26:37 +00:00
henrike@webrtc.org
573a1b45b5 Android: Fixes crash when exiting WebRTCDemo. 
BUG=2738
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:58:06 +00:00
andresp@webrtc.org
f6acf98a46 Fix the android clang bot for compiling with thread annotations. 
TBR=niklas.enbom@webrtc.org
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 21:54:26 +00:00
fischman@webrtc.org
179908c81c JNI Audio: remove dead members. 
BUG=2735
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 23:46:14 +00:00
henrike@webrtc.org
9ee75e9c77 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). 
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 21:42:44 +00:00
kjellander@webrtc.org
f9bdbe3619 Roll chromium_revision 232627:238260 
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003

TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 13:37:12 +00:00
fischman@webrtc.org
d3865e9124 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). 
It is incorrect to wrap close in HANDLE_EINTR on Linux.

BUG=chromium:269623
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4759004

Patch from Mark Mentovai <mark@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 19:10:20 +00:00
henrike@webrtc.org
a750044396 Fixes a crash in VoE when unregistering JNI hooks. 
BUG=11695087
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 22:32:12 +00:00
fischman@webrtc.org
b8cb85b348 Fix broken build on x86 Android 
BUG=2545
R=fischman@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3019004

Patch from Lu Quiang <qiang.lu@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 19:06:08 +00:00
henrike@webrtc.org
05773e5a70 Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. 
TBR=fischman@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2395004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-14 16:25:11 +00:00
henrike@webrtc.org
f53622d42e WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. 
BUG=2083
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-11 21:28:26 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. 
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:23:13 +00:00
henrike@webrtc.org
f8f78b1316 Android OpenSL: Fixes faulty assertion in jni-code. 
BUG=2452
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2342004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 18:41:06 +00:00
henrike@webrtc.org
1fdc51ae2a APK for opensl loopback. 
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2212004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 14:58:19 +00:00
henrike@webrtc.org
c8dea6a00f Use the native sample rate for OpenSL recording. 
BUG=N/A
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2219005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 18:44:51 +00:00
henrike@webrtc.org
6138c5cfa4 OpenSl: fixes crashes externally reported in issue 2361 and 2362. 
BUG=2361,2362
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 18:50:06 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android. 
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 18:24:07 +00:00
braveyao@webrtc.org
c028ee2bf2 Android audio opensles: random deadlock in stopRecording(). 
BUG=2201
Test=WebRTCDemo

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-22 03:14:34 +00:00
henrike@webrtc.org
c92781737c OpenSl bug: not matching playout and record sample rate led to high or low pitch audio (depending on if playout rate was higher or lower than record rate). 
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2031004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 14:13:13 +00:00
pbos@webrtc.org
811269df40 Include files from webrtc/.. paths in audio_device/. 
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 13:24:38 +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
fischman@webrtc.org
546c91dc2e Build all java files into jar for each module on Android 
BUG=None
TEST=All java files in each module are built into jar and used by WebRTCDemo app
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1696004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-01 17:52:39 +00:00
fischman@webrtc.org
dd97ef4e28 Revert 4211 "Build all java files into jar for each module on An..." 
Reason for revert: behold the meltdown of the "trunk" bots on http://build.chromium.org/p/chromium.webrtc.fyi/waterfall

Turns out that include in gyp is fraught with peril: https://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files

> Build all java files into jar for each module on Android
>
> BUG=
> R=fischman@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1636004
>
> Patch from Jeremy Mao <yujie.mao@intel.com>.

TBR=fischman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1660005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 17:39:29 +00:00