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Author SHA1 Message Date
bjornv@webrtc.org
a3b5673879 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.

BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 12:11:20 +00:00
pbos@webrtc.org
1e019d10b8 Fix delivery error-checking missed in r6151. 
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.

BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:38:45 +00:00
solenberg@webrtc.org
57e060251a Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. 
Flakiness was caused by a race condition between two atomic integers shared by two threads. Fixed by counting bad packets (those not containing the expected extension) instead of the good packets.

The CL also eliminates another possible flake by introducing a test fixture which doesn't automatically start sending audio packets when constructed.

BUG=3340,3356
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:27:09 +00:00
andresp@webrtc.org
60015d27ae Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} 
This allows use of webrtc field trials  and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.

Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.

BUG=crbug/367114
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 09:39:51 +00:00
bjornv@webrtc.org
1b21a57902 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 
Macro was only mapping a function used in one place.

BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 06:40:31 +00:00
bjornv@webrtc.org
d83d607271 common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED 
* Moved the macro to randomization_functions and made it static const.
* Made WebRtc_IncreaseSeed() static, since it is not used outside this function.
* Style guide changes.

BUG=3348,3353
TESTED=trybots, common_audio_unittests, modules_unittests, modules_tests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 06:38:47 +00:00
wu@webrtc.org
75718cf80a * Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly. 
* Updated to ClockTest.NtpTime to verify the returned NTP is at least larger than kNtpJan1970.

Current implementation uses timeGetTime, which returns the time since windows is started, which can't be converted to NTP time.

BUG=3325
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 23:54:14 +00:00
henrike@webrtc.org
bf58a75dd9 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 21:45:09 +00:00
henrike@webrtc.org
508795f088 Made the presubmit script accept license headers back to 2003 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 18:21:17 +00:00
henrike@webrtc.org
cfdf420e21 Rebase webrtc/base 6129:6163 (svn diff -r 6129:6163 http://webrtc.googlecode.com/svn/trunk/talk/base apply diff manually) 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 16:33:04 +00:00
buildbot@webrtc.org
6bfd6196ff (Auto)update libjingle 67052073-> 67134648 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 16:15:59 +00:00
pbos@webrtc.org
6aeeac95ca Fix Windows debug compile of overrides/ logging. 
Compile error detected when trying to roll to chromium. Adding a cast
of base::PlatformThread::CurrentId() to base::subtle::Atomic32 to match
types in DCHECK_EQ().

See logging.cc error in:
http://build.chromium.org/p/tryserver.chromium/builders/win_chromium_compile_dbg/builds/19944/steps/compile%20%28with%20patch%29/logs/stdio

R=mflodman@webrtc.org, perkj@webrtc.org
TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 13:56:56 +00:00
mflodman@webrtc.org
d5da25063c Revert "Revert "Audio processing: Feed each processing step its choice 
of int or float data"

This reverts commit 6142.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 11:17:21 +00:00
pbos@webrtc.org
024e4d5c6e Fix Win VideoSendStream::...::ToString() compiles. 
Removed an extra ::VideoSendStream in the method declarations.

BUG=3171
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 10:03:24 +00:00
pbos@webrtc.org
1e92b0a93d Add ToString() to VideoSendStream::Config. 
Adds ToString() to subsequent parts as well as a common.gyp to define
ToString() methods for config.h. VideoStream is also moved to config.h.

BUG=3171
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 09:35:06 +00:00
bjornv@webrtc.org
1aae6bf735 common_audio: Removes unused macros 
* WEBRTC_SPL_MUL_32_32_RSFT32BI
* WEBRTC_SPL_IS_NEG

BUG=3348
TESTED=trybots, common_audio_unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 07:22:53 +00:00
henrik.lundin@webrtc.org
b4e80e095f Re-enable almost all NetEqDecodingTests for Android 
All but three tests in NetEqDecodingTest could be re-enabled without
any changes. Also making sure that the TestNetworkStatistics test exits
on first diff. (Otherwise, the log output gets flooded with error
messages.)

The tests that are still disabled are:
NetEqDecodingTest.TestBitExactness
NetEqDecodingTest.TestNetworkStatistics
NetEqDecodingTest.DecoderError

BUG=3343
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 07:14:00 +00:00
braveyao@webrtc.org
7cb4752184 WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. 
This cl is to teach videocapture android how to deinitialize and allow it to be re-initializable.

BUG=3284
TEST=ManualTest
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 03:18:15 +00:00
wu@webrtc.org
54231f0662 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. 
BUG=crbug/371714
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 23:06:23 +00:00
mallinath@webrtc.org
bb6201ae4b TCP remote socket address should have both server hostname and IP address. 
Hostname is necessary when we are creating TLS based socket, for certificate
verification.

BUG=https://code.google.com/p/chromium/issues/detail?id=306285
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 22:43:05 +00:00
fischman@webrtc.org
a150bc9bbf PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine. 
Enables applications that don't want to pay the init/startup cost or request
extra permissions (e.g. audio-only app, or DataChannel-only app).

BUG=3234

Review URL: https://webrtc-codereview.appspot.com/15489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 22:00:50 +00:00
buildbot@webrtc.org
ef5a752c29 (Auto)update libjingle 67043374-> 67044055 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 21:35:19 +00:00
buildbot@webrtc.org
3e924683d4 (Auto)update libjingle 67037200-> 67043374 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 21:29:04 +00:00
jiayl@webrtc.org
4f5801494d Drop the DataChannel message if it's received when the channel is not open. 
It may happen when the JS has closed the channel on the signaling thread while messages are received on the worker thread and posted before the state change is pushed to the worker thread.

BUG=crbug/363005
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 20:32:35 +00:00
buildbot@webrtc.org
372701a872 (Auto)update libjingle 67023528-> 67036361 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 20:27:59 +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
buildbot@webrtc.org
688ed699e0 (Auto)update libjingle 67017551-> 67023528 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 18:26:09 +00:00
henrike@webrtc.org
c50bf7cbd0 Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 18:24:13 +00:00
kjellander@webrtc.org
3147b97f8c LSan suppressions for libjingle tests (fix) 
Apparently wildcards are needed for templates
in the call stack for matching suppressions.

TBR=wu@webrtc.org
BUG=2528, 3345
TEST=None, since I cannot reproduce the leaks on the bot
in my local environment (and we don't have a trybot for this yet).

Review URL: https://webrtc-codereview.appspot.com/18459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 18:05:15 +00:00
kjellander@webrtc.org
7c0f6e1b47 LSan suppressions for libjingle tests (more) 
Additional suppressions after a new test run was completed
after https://webrtc-codereview.appspot.com/18429006 was
submitted.

TBR=wu@webrtc.org
BUG=2528, 3345
TEST=None, since I cannot reproduce the leaks on the bot
in my local environment (and we don't have a trybot for this yet).

Review URL: https://webrtc-codereview.appspot.com/14509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 17:46:23 +00:00
fischman@webrtc.org
2c98af7935 PeerConnection(Java): auto-WrapCurrentThread() when creating PeerConnectionFactory. 
Various pieces of talk/ assume that the current Thread is ThreadManager'd
without checking this, so unconditionally wrap the caller's thread in case it
was created by Java code unbeknownst to ThreadManager.

BUG=2947
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 17:33:32 +00:00
kjellander@webrtc.org
a70dff4427 LSan suppressions for libjingle tests. 
Adding a few more suppressions to make the tests pass
LSan as it's about to be enabled in the main waterfall
at the ASan bot.

TBR=wu@webrtc.org
BUG=2528, 3345
TEST=None, since I cannot reproduce the leaks on the bot
in my local environment (and we don't have a trybot for this yet).

Review URL: https://webrtc-codereview.appspot.com/18429006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6153 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 17:12:01 +00:00
wu@webrtc.org
88abf11cad Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. 
BUG=3111
TEST=try bots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 16:53:51 +00:00
pbos@webrtc.org
4e545cc244 Update webrtcvideoengine2.cc to use DeliveryStatus. 
talk/ changes corresponding to https://review.webrtc.org/12289005/.

BUG=3228
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 13:58:13 +00:00
pbos@webrtc.org
caba2d2a37 Add DeliveryStatus enum to DeliverPacket(). 
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.

R=mflodman@webrtc.org
BUG=3228

Review URL: https://webrtc-codereview.appspot.com/12289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 13:57:12 +00:00
andresp@webrtc.org
581e2172af Fix libjingle to provide a field_trial implementation. 
This completes https://webrtc-codereview.appspot.com/14489004/ by updating libjingle rules.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 13:12:45 +00:00
kjellander@webrtc.org
01edf2e2af Updating LSan third party suppressions. 
Several new third party suppressions have been updated in
Chromium's suppressions file:
https://code.google.com/p/chromium/codesearch#chromium/src/tools/lsan/suppressions.txt
These will solve some of the errors we're seeing at
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20LSan%20%28and%20ASan%29/
which needs to be resolved before switching our
ASan bot to recipes (since the recipe ASan configuration
has LSan enabled by default).

BUG=2527,2528,3345,3346
TEST=Successfully ran the following tests under ASan+LSan locally:
libjingle_media_unittest
libjingle_p2p_unittest
libjingle_peerconnection_unittest
libjingle_unittest

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 12:28:38 +00:00
andresp@webrtc.org
a36ad6929d Add webrtc field trials API. 
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.

Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.

Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 12:24:04 +00:00
henrika@webrtc.org
9f277350f8 Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12299005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 12:04:29 +00:00
henrika@webrtc.org
f383a1b0f2 Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 11:51:45 +00:00
henrik.lundin@webrtc.org
2fa17015d1 Re-enable NetEqExternalDecoderTest for Android 
The test runs without problems now.

BUG=3343
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 11:45:22 +00:00
henrik.lundin@webrtc.org
bf93fb3176 Re-enable NetEQ DecoderDatabase test for Android 
The test was failing because iLBC is not enabled on Android. Now, the
test is using PCM16B instead.

BUG=3343
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 10:42:03 +00:00
mflodman@webrtc.org
b1a66d166c Revert "Audio processing: Feed each processing step its choice of int or float data" 
This reverts r6138.

tbr=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:39:56 +00:00
solenberg@webrtc.org
db60434b31 Re-enable the BitrateEstimatorTest cases for the Call API. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:15:19 +00:00
henrik.lundin@webrtc.org
5c49c64de5 Remove all use of AudioFrame::energy_ from AudioCodingModule 
Since r6117, the energy is always calculated in the mixer module,
regardless of the value that ACM sets for energy_.

This part of the the aftermath of issue 3255.

BUG=3255
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:06:52 +00:00
bjornv@webrtc.org
06c1d6f3a1 VoEVolumeTest: Adds error return tests. 
BUG=367
TESTED=trybots, voe_auto_test
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:03:33 +00:00
kwiberg@webrtc.org
934a265a47 Audio processing: Feed each processing step its choice of int or float data 
Each audio processing step is given a pointer to an AudioBuffer, where
it can read and write int data. This patch adds corresponding
AudioBuffer methods to read and write float data; the buffer will
automatically convert the stored data between int and float as
necessary.

This patch also modifies the echo cancellation step to make use of the
new methods (it was already using floats internally; now it doesn't
have to convert from and to ints anymore).

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the echo canceller no longer unnecessarily
converts float data to int and then immediately back to float for each
iteration in the loop in EchoCancellationImpl::ProcessCaptureAudio.)

BUG=
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:01:35 +00:00
pbos@webrtc.org
3d5cb33da4 Remove WEBRTC_TRACE use in video_capture/ 
Does not touch platform-specific code.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 08:42:07 +00:00
pbos@webrtc.org
4e2806d85f Remove WEBRTC_TRACE uses in video_engine/ 
Complements fixes by mflodman@.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 08:02:22 +00:00
kjellander@webrtc.org
98c76a120d Make vie/voe_auto_test accept non-supported flags without error. 
With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005

BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz

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

Review URL: https://webrtc-codereview.appspot.com/13499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 06:01:40 +00:00