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Author SHA1 Message Date
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
103f3289b5 Fix the binary layout of ProcessThreadImpl. 
We apparently hit an obscure problem on mac where seemingly an unaligned mutex causes memory corruption.
The effect was that the |modules_| list became corrupt and we crashed.  At this point I'm not exactly
sure what the alignment requirements are but for now, I've fixed up the layout in a way that doesn't cause these same issues.

I'm also changing auto->proper type at the request of drive by reviewers from my previous cl in the same file.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38989004

Cr-Commit-Position: refs/heads/master@{#8286}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-08 00:48:40 +00:00
tommi@webrtc.org
1d4830a077 Disable ProcessThread tests that are dependent on timing. 
Some of the bots are too slow for the tests to make much sense as they are.

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39869004

Cr-Commit-Position: refs/heads/master@{#8281}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-07 08:44:40 +00:00
tommi@webrtc.org
bc3241a8cc Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :) 
TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40659004

Cr-Commit-Position: refs/heads/master@{#8262}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 11:28:41 +00:00
tommi@webrtc.org
0c3e12b7bf Revamp the ProcessThreadImpl implementation. 
* Add a new WakeUp method that gives a module a chance to be called back right away on the worker thread.
* Wrote unit tests for the class.
* Significantly reduce the amount of locking.
  - ProcessThreadImpl itself does a lot less locking.
  - Reimplemented the way we keep track of when to make calls to Process.
    This reduces the amount of calls to TimeUntilNextProcess and since most implementations of that function grab a lock, this means less locking.
* Renamed ProcessThread::CreateProcessThread to ProcessThread::Create.
* Added thread checks for Start/Stop.  Threading model of other functions is now documented.
* We now log an error if an implementation of TimeUntilNextProcess returns a negative value (some implementations do, but the method should only return a positive nr of ms).
* Removed the DestroyProcessThread method and instead force callers to use scoped_ptr<> to maintain object lifetime.

BUG=2822
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35999004

Cr-Commit-Position: refs/heads/master@{#8261}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 09:44:45 +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
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory. 
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:30:52 +00:00
pbos@webrtc.org
dd8f6f3d48 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. 
_t names are reserved in POSIX.

BUG=162
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:18:42 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
pbos@webrtc.org
273a414b0e Report encoded frame size in VideoSendStream. 
Implements reporting transmitted frame size in WebRtcVideoEngine2.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033

Review URL: https://webrtc-codereview.appspot.com/33399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:23:21 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
henrike@webrtc.org
cc774a69cb Mark all virtual overrides in the hierarchies of RtpDump and 
VCMPacketizationCallback as such.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also marks all other such overrides in the affected files.

BUG=none
TEST=none
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 22:45:54 +00:00
henrike@webrtc.org
47658f1269 Mark all virtual overrides in the hierarchy of AudioPacketizationCallback, 
RTPStream, and NetEq as such.  Also mark all other virtual overrides in the same
files.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which
was marked pure virtual in the header.  (Pure virtual destructors still need a
definition.)  Because there is another pure virtual method in this class, the
class is already abstract, so there's no benefit to making the desturctor pure.
Making it non-pure allows removing the separate source file.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 22:14:59 +00:00
pbos@webrtc.org
047a46f8b4 Remove Android.mk build files. 
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 08:48:51 +00:00
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

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

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
wuchengli@chromium.org
637c55f45b Add support of texture frames for video capturer. 
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.

- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
     common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 07:00:51 +00:00
wuchengli@chromium.org
89e8ffb395 Revert "Add support of texture frames for video capturer." 
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.

Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.

BUG=chromium:362437
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 14:12:58 +00:00
wuchengli@chromium.org
efe15355ee Add support of texture frames for video capturer. 
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
  are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 12:40:27 +00:00
henrik.lundin@webrtc.org
74767401f2 Fix a bug preventing FilePlayer from playing encoded wav files 
A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).

This CL fixes the issue, and adds a unit test for the FilePlayer.

BUG=3386
R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-26 13:37:45 +00:00
andrew@webrtc.org
21299d4e00 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. 
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.

Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc

Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.

BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 19:00:59 +00:00
henrike@webrtc.org
ceffdbc371 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() 
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:11:07 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 20:08:03 +00:00
stefan@webrtc.org
34c5da6b5e Cleaned up logging in video_coding. 
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.

BUG=3153
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 14:08:35 +00:00
asapersson@webrtc.org
8b2ec15d1e Convert WEBRTC_TRACE to LOG in utility. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 07:59:43 +00:00
braveyao@webrtc.org
4f0801bd39 AviRecorder is missing a critical section. 
BUG=2885
TEST=AUTOTEST
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 09:19:36 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files. 
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 15:21:30 +00:00
andrew@webrtc.org
de7c9e8884 Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. 
Move the logic to common.gypi to reduce the chance of the define being
unprotected in the future.

BUG=b/12018143
TESTED=git try, and local Linux build with -Denable_video=0
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-09 16:23:00 +00:00
andrew@webrtc.org
621df678c8 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. 
Mostly to remove a long-standing TODO...

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 10:27:23 +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
pbos@webrtc.org
e546f02c84 Remove include_dirs from utility. 
BUG=1662
TEST=compile on trybots
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-01 09:29:09 +00:00
andrew@webrtc.org
eb524d997b Remove deprecated AudioCodingModule::Destroy. 
Have Channel hold a pointer rather than reference, and shorten the name.

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 23:02:24 +00:00
andresp@webrtc.org
8fa436bd65 Remove use of vcm->ResetDecoder from modules/utility. 
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2203006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-16 11:26:35 +00:00
andrew@webrtc.org
eda189be14 Remove redundant STR_CASE_CMP macro definitions. 
R=minyue@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2187005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:50:10 +00:00
mikhal@webrtc.org
f1e807c0e5 Removing FrameForStorage 
R=pwestin@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2142004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 22:34:41 +00:00
wu@webrtc.org
9dba525627 * Update libjingle to 50389769. 
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 20:36:57 +00:00
fischman@webrtc.org
f696f253b2 Invert dependency between webrtc_utility and media_file targets to reflect reality. 
BUG=2166
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1953004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 18:45:19 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
pbos@webrtc.org
8b06200802 Include files from webrtc/.. paths in utility/. 
BUG=1662
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1786004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:28:10 +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
kjellander@webrtc.org
fec34d7afa Merge webrtc_utility_unittests into modules_unittests. 
This CL eliminates the webrtc_utility_unittests test target.

NOTICE: Upon committing, this test must be removed from the
Buildbot configuration.

BUG=1843
TEST=trybots passing. Compiled and ran modules_unittests, verified the
AudioFrameOperationsTest test executes and passes.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1584004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 08:58:46 +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
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
pbos@webrtc.org
c75102eba7 WebRtc_Word32 -> int32_t in utility/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:32:55 +00:00
henrika@webrtc.org
4ff956f428 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer 
BUG=159112

Review URL: https://webrtc-codereview.appspot.com/1201007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 11:59:11 +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
tina.legrand@webrtc.org
7a7a008031 Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Committed: https://code.google.com/p/webrtc/source/detail?r=3543

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 10:27:48 +00:00
tina.legrand@webrtc.org
eb7ebf20ed Revert 3543 
> Changing non-const reference arguments to pointers, ACM
> 
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
> 
> BUG=issue1372
> 
> Review URL: https://webrtc-codereview.appspot.com/1103012

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:57:31 +00:00
tina.legrand@webrtc.org
374aa49e1a Changing non-const reference arguments to pointers, ACM 
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.

BUG=issue1372

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-20 15:22:23 +00:00
andrew@webrtc.org
ae1a58bba4 Replace AudioFrame's operator= with CopyFrom(). 
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.

Review URL: https://webrtc-codereview.appspot.com/1031007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 04:44:30 +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