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
buildbot@webrtc.org
cd846dd374
(Auto)update libjingle 66924241-> 66927231
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6134 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 22:58:27 +00:00
buildbot@webrtc.org
da510c5de6
(Auto)update libjingle 66923202-> 66924241
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6132 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 22:30:56 +00:00
fischman@webrtc.org
d8af5b51c0
Deallocate the result of mach_host_self() when done with it, fixing a
...
port leak.
The port rights obtained by mach_host_self() and mach_thread_self() need
to be deallocated with mach_port_deallocate(). They consume finite
system-wide resources. This is in contrast to mach_task_self(), which is
a macro that wraps an extern global variable, and must not be
deallocated.
http://crbug.com/105513 shows the sorts of problems that can occur when
these aren't properly deallocated.
R=fischman@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15469004
Patch from Mark Mentovai <mark@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6131 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 22:18:48 +00:00
buildbot@webrtc.org
c14f521b1b
(Auto)update libjingle 66887616-> 66900106
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6130 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 18:52:57 +00:00
henrike@webrtc.org
f048872e91
Adds a modified copy of talk/base to webrtc/base. It is the first step in
...
migrating talk/base to webrtc/base.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 18:00:26 +00:00
buildbot@webrtc.org
3e01e0b16c
(Auto)update libjingle 66867790-> 66887616
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 17:54:10 +00:00
henrike@webrtc.org
c156174da8
Suppressing all tests for WebRtcVideoEngine2 for Win DrMemory Full.
...
BUG=3336
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6124 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 16:47:32 +00:00
bjornv@webrtc.org
8d63d0ee70
Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255
...
Rewritten the test to only check for valid volume when we have actually received a value from the audio device. To check if we have actually received a volume value is out of the scope for this test.
BUG=webrtc:367
TESTED=trybots
R=tina.legrand@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6123 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 14:14:56 +00:00
andresp@webrtc.org
93ec9c557b
Revert "FieldTrial implementation for webrtc." (rev 6089)
...
New wiring plans require it to be landed first in chrome for a cleaner roll of webrtc.
BUG=crbug/367114
R=tommi@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6122 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 14:09:40 +00:00
asapersson@webrtc.org
e41dbee8a6
Reduced kMaxSampleDiffMs (limit to 22fps).
...
BUG=1577
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6121 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 13:45:13 +00:00
pbos@webrtc.org
023b101f4e
Move gflags usage to video_loopback.
...
gflags aren't used by the test environment and is an unnecessary
dependency. They're only used by the video_loopback target, so moving
them there.
R=mflodman@webrtc.org
BUG=3113
Review URL: https://webrtc-codereview.appspot.com/12379006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6120 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:26:40 +00:00
pbos@webrtc.org
b5a22b1464
Revert r6110 and r6109.
...
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.
BUG=
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:07:01 +00:00
henrik.lundin@webrtc.org
c3e8abda7c
Deleting all NetEq3 files
...
NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14469007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 10:40:52 +00:00
henrik.lundin@webrtc.org
4d363ae305
The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.
...
This part of the the aftermath of issue 3255.
BUG=3255
R=andrew@webrtc.org , henrike@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6117 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:50:02 +00:00
perkj@webrtc.org
e9a604accd
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
...
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.
http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457
> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
>
> BUG=N/A
> R=andrew@webrtc.org , wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12199004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:15:48 +00:00