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Author SHA1 Message Date
tkchin@webrtc.org
9343cf67a9 Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS. 
BUG=3581
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 17:13:28 +00:00
tkchin@webrtc.org
122caa51b1 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. 
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:20:47 +00:00
tommi@webrtc.org
d212ffcfc6 Remove unnecessary build message. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 11:15:35 +00:00
phoglund@webrtc.org
241a9b0b65 Fixing compile error. 
Made a mistake in https://webrtc-codereview.appspot.com/13849004/,
fixing that here.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 11:48:37 +00:00
phoglund@webrtc.org
22292df53b Adding explicit check for using dummy file devices. 
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 11:39:19 +00:00
tkchin@webrtc.org
74bf7a6523 Add tkchin@ to OWNERS. 
Adding myself to OWNERS of subdirectories containing iOS bits.  Added niklas.enbom@ for audio_device and wu@ for everything else.

R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:38:28 +00:00
kjellander@webrtc.org
1227ab89a7 GN: Add BUILD.gn files + kjellander to OWNERS 
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.

I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.

I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.

BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

R=brettw@chromium.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-23 19:21:07 +00:00
kjellander@webrtc.org
a1bfc50a72 Pass GYP DEPTH variable to isolate. 
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.

Also update all our .isolate files to use the <(DEPTH)
variable.

BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.

R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:02:15 +00:00
phoglund@webrtc.org
8454ad1b3e Reland: Making WebRTC able to play and record audio to files for tests. 
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 14:12:04 +00:00
minyue@webrtc.org
e08a11c4a1 Revert 6395 "Making WebRTC able to play and record audio to file..." 
> Making WebRTC able to play and record audio to files for tests.
> 
> By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
> WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
> play out audio to a file and feed audio in from a file. We want to do
> so we can better test WebRTC-using applications by recording what the
> audio stack outputs and feeding known audio in for quality tests.
> 
> R=henrika@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/20609004

TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 10:40:30 +00:00
phoglund@webrtc.org
fa042ca15d Making WebRTC able to play and record audio to files for tests. 
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 09:57:23 +00:00
kjellander@webrtc.org
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate 
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:42:53 +00:00
wu@webrtc.org
94454b71ad Fix the chain that propagates the audio frame's rtp and ntp timestamp including: 
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 20:34:08 +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
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 21:18:46 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..." 
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 11:07:29 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:20:44 +00:00
wu@webrtc.org
cb711f77d2 Add interface to propagate audio capture timestamp to the renderer. 
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:39:11 +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
d59359af4d Remove 44.1 kHz workaround from the iOS AudioDevice. 
Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.

This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.

BUG=1395
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 18:07:49 +00:00
fischman@webrtc.org
ca539bbed0 iOS: baby steps to being able to include_tests=1 
- pull iossim in DEPS even when on mac (because bug 2152)
- fix audio_device_test_api.cc's use of bool instead of bool* (!)
- move unused-on-mobile message to non-mobile-only section of
  hardware_before_streaming_test.cc

BUG=3185
R=kjellander@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 20:26:41 +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
xians@webrtc.org
5692531f18 Added a new OnMoreData() interface which will not feed the playout data to APM. 
BUG=3147
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 10:50:37 +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
fischman@webrtc.org
a789f3720a VoiceEngine(iOS & Android): removed NOT_SUPPORTED 
Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
  bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds

BUG=2050,3132
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 00:16:35 +00:00
pbos@webrtc.org
0e65fdaa3b Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. 
BUG=chromium:346399
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10139004

Patch from Peter Kasting <pkasting@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-21 10:26:42 +00:00
fbarchard@google.com
66061992fb ifdef the alsa code based on macro USE_X11 
BUG=none
TEST=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 03:05:05 +00:00
xians@webrtc.org
c1e28038ba Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-02 15:30:20 +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
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
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
fischman@webrtc.org
000dde99c8 Android build: make it quiet on success and not overly noisy on failure. 
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 22:49:35 +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
andresp@webrtc.org
7fb75ecbd4 Add thread_annotations for clang targets. 
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.

R=niklas.enbom@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 20:20:50 +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
7ae8495779 Removed unnecessary Pulse init from VoE startup. 
Saves 10% (~260ms) of the total PeerConnectionTest wallclock time.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-10 21:01:34 +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
stefan@webrtc.org
b082ade3db Hook up audio/video sync to Call. 
Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:45:11 +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
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
kjellander@webrtc.org
3555303cb0 Roll chromium_revision 226126:228675 and fix clang warnings 
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.

This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.

TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 20:10:17 +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
kjellander@webrtc.org
3f9288f987 Add APK and isolate target for video_engine_tests 
Add .isolate file and _run target for video_engine_tests.

Move tools/swarm_client to be untracked in all .isolate file,
so refactorings in swarm_client doesn't require us updating
all our .isolate files (similar to the changes for the
Chromium tests done in:
https://src.chromium.org/viewvc/chrome?view=rev&revision=218844)

Update modules_unittests.isolate with new NetEq4 reference files
needed.

TEST=trybots passing
I also setup a Chromium workspace where I patched third_party/webrtc
with the changes in this CL, followed by compiling with the settings
described in
https://code.google.com/p/webrtc/issues/detail?id=1882#c11
I then verified that the video_engine_tests_apk dir was created
in the output folder.
BUG=1916,2462
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 18:20:38 +00:00
henrike@webrtc.org
ad2eb6f67d Unbreaks Android build after r4915. 
TBR=ajm@webrtc.org

BUG=Not filed

Review URL: https://webrtc-codereview.appspot.com/2348005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-04 14:21:23 +00:00