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Author SHA1 Message Date
henrik.lundin@webrtc.org
fdf2053787 Remove AudioCodingModuleFactory 
These were no longer used anywhere in the code.

BUG=2996
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:22:14 +00:00
henrik.lundin@webrtc.org
0bc9b5a5a7 Add clock to ACM config struct 
The purpose is to clean up the ACM interface a bit. This is a
follow-up of a comment in http://review.webrtc.org/13379004/.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:09:31 +00:00
bjornv@webrtc.org
059488f2ea AEC: Startup phase only runs if reported_delay_enabled 
TESTED=trybots, modules_unittests
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:09:26 +00:00
buildbot@webrtc.org
af6640fce7 (Auto)update libjingle 65729829-> 65752960 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 21:31:51 +00:00
fischman@webrtc.org
874920f22e Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. 
The C (unoptimized) version of this function takes less than 1% of NetEq4 and
has been crashing since the NetEq3->4 move.

BUG=3243
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6003 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 20:09:09 +00:00
fischman@webrtc.org
82a045aae0 APM: limit native sample rate to 16kHz on mobile. 
Required by AECM which assert-fails on higher sample rates.

BUG=3259
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 17:26:32 +00:00
fischman@webrtc.org
f27fdeb9c9 AppRTCDemo(android): don't initialize process-globals more than once. 
BUG=3257
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 16:32:38 +00:00
michaelbai@google.com
497ff21fad Using realpath instead of android_src in Android webview 
BUG=367235
R=andrew@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/20369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 15:46:46 +00:00
andrew@webrtc.org
1152fe2247 Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. 
BUG=b/13171670
R=fbarchard@google.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5999 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 15:46:32 +00:00
pbos@webrtc.org
de1429e9ad Add thread annotations to Call API. 
Also constified a lot of pointers and reordered members to make
protected members more grouped together.

R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770

Review URL: https://webrtc-codereview.appspot.com/15399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 13:00:21 +00:00
kjellander@webrtc.org
7d825e9b2c Revert "libjingle_unittest now compiles and passes on iOS!" 
This reverts commit r5986 as it fails compilation on Mac
(non-iOS). The failure was not discovered on the commitbots
since they don't clobber their builds.

BUG=3241
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 12:50:47 +00:00
pbos@webrtc.org
c891577e6d Disable flaky CaptureNtpTimeWithNetworkJitter. 
TBR=wu@webrtc.org
BUG=3271

Review URL: https://webrtc-codereview.appspot.com/19389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 11:57:32 +00:00
bjornv@webrtc.org
494aa0e93d AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h 
These enums are noly used internally in aec_core.c and it makes more sense to put them in aec_core_internal.h

TESTED=trybots
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 11:42:27 +00:00
pbos@webrtc.org
8dfe8ff590 Disable capture test for FrameRate on Windows. 
Flaky on Windows, has been for a while.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=3270

Review URL: https://webrtc-codereview.appspot.com/19389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 11:27:36 +00:00
henrik.lundin@webrtc.org
e772c71743 Introduce a config struct for AudioCoding module 
The config struct currently contains the module ID, and the NetEq
config struct, but will be extended in the future. The purpose of this
change is to expose certain NetEq settings to the ACM interface.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 10:16:57 +00:00
pbos@webrtc.org
f043f79711 Disabling flaky CanReceiveFec. 
CanReceiveFec is flaky, likely to the test expecting the first FEC
packet to always be decoded and rendered.

R=stefan@webrtc.org
BUG=3269

Review URL: https://webrtc-codereview.appspot.com/19379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 09:00:50 +00:00
pbos@webrtc.org
69e9950469 Disable flaky RunsRtpRtcpTestWIthoutErrors. 
BUG=1790
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:49:07 +00:00
henrik.lundin@webrtc.org
12a34247a4 Fix the NetEq build 
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:36:35 +00:00
henrik.lundin@webrtc.org
116ed1d4f0 Include buffer size limits in NetEq config struct 
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.

The old constants governing the packet buffer limits are deleted.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:20:04 +00:00
henrik.lundin@webrtc.org
b08bbf57a6 Add henrik.lundin as owner in AudioCoding module 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:15:35 +00:00
mallinath@webrtc.org
a0d3067575 Use CreatePeerConnection method which accepts port_allocator. 
Other method will be removed, in a different CL.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-26 00:00:15 +00:00
fischman@webrtc.org
95cd1551f8 libjingle_unittest now compiles and passes on iOS! 
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && ~/src/ios-deploy/ios-deploy -d -u -v -b ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app
Note that the test's use of signals means that lldb will break in the middle of the suite.  To ignore these signals tell lldb:

pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue

BUG=3241
R=noahric@google.com, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:59:56 +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
buildbot@webrtc.org
658a94595d (Auto)update libjingle 65619249-> 65622932 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 22:03:30 +00:00
buildbot@webrtc.org
ff90ed6e96 (Auto)update libjingle 65561104-> 65619249 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 21:12:10 +00:00
andrew@webrtc.org
2eceb8ef46 Roll third_party/opus 258909:262302 
Suppresses the nanny "not-optimized" warning.

TBR=minyue
BUG=2864

Review URL: https://webrtc-codereview.appspot.com/14379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 15:29:33 +00:00
stefan@webrtc.org
0175d76c72 Fix leak in remote bitrate estimator tests introduced in r5980 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 11:38:57 +00:00
stefan@webrtc.org
4f616a02bd Support for simulating multiple independent flows in a network. 
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 10:59:24 +00:00
asapersson@webrtc.org
46106f2a05 Casting char to int in logs. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 07:02:52 +00:00
buildbot@webrtc.org
2b93402e36 (Auto)update libjingle 65484212-> 65561104 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 00:18:27 +00:00
jiayl@webrtc.org
cc1ba15fe7 Returns a NULL frame on all platforms if the captured window is closed. 
Part of the fix for crbug/360181.
On Mac/Linux, it previously continues capturing even if the window is closed.
Now it stops by returning a NULL frame.
On Windows, it used to stop capturing when the window is minimized. Now fixed to match other platforms.
Note: the crbug still needs a chrome side fix to close the notification bar.
This fix only stops the stream (i.e. stream onended event fired).

BUG=crbug/360181
TESTED=manually tested in Chrome
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 23:45:56 +00:00
wu@webrtc.org
cd70119a10 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. 
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 22:10:24 +00:00
wu@webrtc.org
93fd25c20c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. 
* Cast rtp header extension to int in log in rtp_utility.cc.

BUG=3237
TEST=try bots
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 20:33:08 +00:00
henrik.lundin@webrtc.org
439a4c49f9 Add an output capacity parameter to ACMResampler::Resample10Msec() 
Also adding a unit tests to make sure that a desired output frequency
of 0 passed to AudioCodingModule::PlayoutData10Ms() is invalid.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 19:05:33 +00:00
andrew@webrtc.org
103657b484 Add keyboard channel support to AudioBuffer. 
Also use local aliases for AudioBuffers for brevity.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 18:28:56 +00:00
henrik.lundin@webrtc.org
d57b8149c2 Fix the Android compilation (better structure for NetEq test libs) 
This change should make the Android targets compile again. The reason
for the failure was a highly dubious structure in the gypi files. With
this fix, the structure is somewhat cleaner. Still room for improvement.

BUG=3254
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 13:19:04 +00:00
pbos@webrtc.org
5ca6a5387e Remove TraceCallback use from Call. 
Non-global logging isn't supported, and having a per-call logging
dispatch seems over-eager and adds more complexity than it's worth. The
current implementation is racy on initialization due to missing atomics
support. Besides, logging support should be separate from use of Call.

R=mflodman@webrtc.org
BUG=3250,3157

Review URL: https://webrtc-codereview.appspot.com/12329006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 11:35:33 +00:00
pbos@webrtc.org
a5c8d2c9b3 Rename Start/Stop in Video{Send,Receive}Streams. 
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.

R=mflodman@webrtc.org
BUG=3227

Review URL: https://webrtc-codereview.appspot.com/12329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 11:13:21 +00:00
henrik.lundin@webrtc.org
0a2277448e Fixing a bug in ACM2 where the output frame energy was incorrectly set 
The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).

Implemented a unit test to verify ACM output frame.

BUG=3255
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 08:11:39 +00:00
andrew@webrtc.org
f26c9e8369 Use unique filenames in AudioProcessingTests for parallelization. 
TBR=bjornv
TESTED="gtest-parallel -w 32 --gtest_filter=*AudioProcessingTests*
out/Debug/modules_unittests" passes.

Review URL: https://webrtc-codereview.appspot.com/14369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 03:46:46 +00:00
buildbot@webrtc.org
3f1aa24078 (Auto)update libjingle 65469804-> 65484212 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 00:00:12 +00:00
jiayl@webrtc.org
0d915ff603 Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet. 
R=juberti@webrtc.org, mallinath@webrtc.org

BUG=3244

Review URL: https://webrtc-codereview.appspot.com/12299006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 20:46:29 +00:00
bjornv@webrtc.org
e9d3760d5c AEC: Adds a reported_delay_enabled_ flag 
Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.

TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 13:20:07 +00:00
henrik.lundin@webrtc.org
26e2b687fc Remove ACM1/ACM2 switching from VoiceEngine tests 
The option to run VoiceEngine tests with both ACM1 and ACM2 was
introduced while the two versions of AudioCoding module where both
in use. Now, ACM1 is being deprecated, and the tests should use the
defualt one (ACM2).

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 08:39:41 +00:00
andrew@webrtc.org
db144429b2 Exclude the new AudioProcessingTest from some sanitizer bots. 
It takes too long.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/12359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 04:16:24 +00:00
andrew@webrtc.org
46b31b17df Restore sample_rate_hz() until Chromium is updated to not use it. 
TBR=bjornv
TESTED=Chromium builds against webrtc head.
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 03:33:54 +00:00
buildbot@webrtc.org
504fc89f36 (Auto)update libjingle 65394435-> 65417850 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 03:23:19 +00:00
tkchin@webrtc.org
19b1be159e Provide GetStats method in RTCPeerConnection 
BUG=3144
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 21:05:38 +00:00
andrew@webrtc.org
ddbb8a2c24 Support arbitrary input/output rates and downmixing in AudioProcessing. 
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.

- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 21:00:04 +00:00
henrik.lundin@webrtc.org
34fe0153b9 Reland "Stop using ACM factory in VoiceEngine" 
This change was originally landed as r5954, but had to be reverted in
r5955 due to bots failing. The failures should be fixed in r5956,
so the original change is now relanded.

BUG=2996
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 19:04:34 +00:00