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Author SHA1 Message Date
minyue@webrtc.org
33ccdfa1f5 Relanding r7807. 
r7807 was reverted to be excluded from the cause of a failure.

It has been verified and can reland now.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 12:14:12 +00:00
minyue@webrtc.org
52bc4f4797 Revert 7807 "Removing unused opus wrapper APIs." 
> Removing unused opus wrapper APIs.
> 
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
> 
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
> 
> BUG=
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28139004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 11:00:50 +00:00
minyue@webrtc.org
e54a6342dd Removing unused opus wrapper APIs. 
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().

WebRtcOpus_DecodePlcMaster/Slave() are also removed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 08:47:25 +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
marpan@webrtc.org
4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff 
Pick up the libvpx roll: https://codereview.chromium.org/674753002

Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3

Clang is not updated in this roll.

Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')

Update rate control parameter in vp9 test.

R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/23229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 20:10:26 +00:00
pbos@webrtc.org
c86e45d7c4 Fix parallelizability in modules_tests. 
R=henrik.lundin@webrtc.org
BUG=3873
TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests

Review URL: https://webrtc-codereview.appspot.com/24799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 10:05:40 +00:00
minyue@webrtc.org
60fbd65482 Removing error triggered for disabling FEC on non-opus 
A failure was triggered when one sets FEC status on a codec that does not support FEC. While it is definitely logical when one wants to enable it, it makes no good sense if one tries to disable it.

BUG=
R=tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 14:36:30 +00:00
henrik.lundin@webrtc.org
741711a861 Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. 
r7049 added some unnecessary casts ("return 0" -> "return static_cast<uint16_t>(0)").  r7123 converted these to "return 0u".  The original impetus for this was to eliminate type conversion warnings.  However, the 'u's are unnecessary; Visual Studio can return "0" from a function returning an unsigned value without producing a warning.  The real reason for the original warnings was that the code was returning -1 from a function returning an unsigned value, which does need a cast; the -1s were eliminated before the above two revisions landed.

Also reverse the order of some conditionals to prevent possible underflow.
While the underflow wouldn't have changed the behavior of the code, it's easier
to reason about the code when such underflow can't happen, and possibly safer
against future modifications as well.

BUG=3663
TEST=none
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 07:38:14 +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
fbarchard@google.com
a941970d4a Change explicit static cast from int to uint16_t to implicit cast of 0u. 
BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:37:27 +00:00
andresp@webrtc.org
262e676a08 Reland rev 7041 with BUILD.gn files. 
Original description:
  Audio codecs to include webrtc/typedefs.h

  Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

  CL Generated with:
  $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:28:48 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths. 
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

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

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 09:55:40 +00:00
fbarchard@google.com
9328f39a3e cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error 
BUG=3663
TESTED=ninja local build on windows.
R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 23:05:07 +00:00
henrike@webrtc.org
1b8b4c4959 Revert 7041 " Audio codecs to include webrtc/typedefs.h" 
Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio

R=turaj@webrtc.org
TBR=andresp@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/19219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 19:42:16 +00:00
andresp@webrtc.org
9730d3aae9 Audio codecs to include webrtc/typedefs.h 
Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 14:37:18 +00:00
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
minyue@webrtc.org
f563e85ab0 This is to re-open an earlier CL 
https://webrtc-codereview.appspot.com/16619005/

which is reverted due to an issue in audio conference mixer.

This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/

BUG=webrtc:3155
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:11:27 +00:00
minyue@webrtc.org
d42da54768 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." 
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
> 
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16619005

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 09:50:12 +00:00
minyue@webrtc.org
8f8503d947 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:02:05 +00:00
tina.legrand@webrtc.org
65d61c3924 Opus send rate overflows if over 65 kbps 
The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 13:42:51 +00:00
minyue@webrtc.org
aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

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

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 15:16:51 +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
henrik.lundin@webrtc.org
adaf809612 Removing AudioCoding duplicate tests 
Reverting to using one version of ACM in ACM tests.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 08:29:10 +00:00
tina.legrand@webrtc.org
92c0e29963 Run Opus with lower complexity setting on Android, iOS and/or ARM 
This CL includes a call to Opus to set a lower complexity figure, if we are compiling for Android, iOS, or ARM (e.g. ChromeOS on ARM), where we know the devices are not powerful enough to run on higher complexity setting.

BUG=3093
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 14:38:36 +00:00
tina.legrand@webrtc.org
ba5a6c3d89 ACM2/NetEq4 did not decode Opus in stereo 
Two problems fixed in this CL:
- setting Opus decoder to stereo had no effect, and decoding always generated mono audio
- changing decoding setting from mono to stereo, or stereo to mono, for OPUS also had no effect (but required another change than the first one).

BUG=3082
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-23 09:58:48 +00:00
henrik.lundin@webrtc.org
3ab57c514c Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets 
This is a relanding of r5725, now with a fix for the failing tests.

BUG=2935
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-20 15:09:38 +00:00
andrew@webrtc.org
21df84711a Disable TestOpusNewACM on Android. 
It crashes flakily.

TBR=tlegrand
BUG=3006

Review URL: https://webrtc-codereview.appspot.com/9809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-11 20:40:59 +00:00
turaj@webrtc.org
78f0db4710 Fix the break caused by r5579. 
TBR=tlegrand@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/8939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 23:07:31 +00:00
turaj@webrtc.org
c2d69d3229 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. 
BUG=2944
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 20:31:17 +00:00
tina.legrand@webrtc.org
056287eee0 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. 
BUG=issue2874
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 13:45:54 +00:00
andresp@webrtc.org
d0b436a935 Revert "Activate ACM test for Android in modules_tests." (rev5364). 
TBR=turaj@webrtc.org,tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 13:15:59 +00:00
turaj@webrtc.org
7cc64b3747 Activate ACM test for Android in modules_tests. 
TEST=local on Nexus 7.
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:35:09 +00:00
turaj@webrtc.org
7a05ae5c69 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. 
The main() was deleted in r4731.

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

Review URL: https://webrtc-codereview.appspot.com/2370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 18:16:53 +00:00
minyue@webrtc.org
3e427263ee Reducing opus_test runtime to pass Android test 
BUG=2609
R=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 22:03:52 +00:00
turaj@webrtc.org
55e1723713 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. 
BUG=2515
TEST=reproduced locally on linux and verified the fix resolves the issue.
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 04:40:09 +00:00
turaj@webrtc.org
6ea3d1cc9e ACM test are modified to run with both ACM1 and ACM2. 
Beside the changes in test files. acm2/acm_generic_codec.cc and acm2/audio_coding_module_impl.cc are modified to fix a bug.

Also, nack{.cc, .h, _unittest.cc} are removed form main/sourc as nack files in both ACM1 and ACM2 are essentially identical.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2192005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 21:44:33 +00:00
dwkang@webrtc.org
63fe8e1f38 Enable SetInitialPlayoutDelay on Android. 
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.

BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release  -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 05:42:22 +00:00
turaj@webrtc.org
532f3dc548 Compile ACM2 and ACM1. 
First patch set is the same as patch set 3 of http://review.webrtc.org/2237004/

-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these
APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these
APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF
detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded.
Remove dynamic payload-type assignment.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4785 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 00:12:23 +00:00
stefan@webrtc.org
1c77dfd521 Revert r4772 "Compile ACM1 and ACM2." 
Breaks Android build.

TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2244004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 12:34:05 +00:00
turaj@webrtc.org
367baa6eb3 Compile ACM1 and ACM2. 
-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded. Remove dynamic payload-type assignment.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2237004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 00:36:11 +00:00
turaj@webrtc.org
48af652ea5 Prepare to compile ACM1 and ACM2. 
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 23:06:59 +00:00
andrew@webrtc.org
89df092807 Make the destructor of AudioCodingModule public. 
This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:27:43 +00:00
henrike@webrtc.org
256b83146c Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. 
BUG=2364
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 20:43:13 +00:00
mflodman@webrtc.org
65abb6b1ed Revert 4671 "Enable SetInitialPlayoutDelay on Android." 
Tests enabled in r4671 failed:
build.chromium.org/p/client.webrtc/builders/Android%20Tests/builds/31/steps/slave_steps/logs/stdio

> Enable SetInitialPlayoutDelay on Android.
> 
> Background:
> In Chrome mirroring which uses 500ms buffering mode,
> audio video mismatch happens in the begining because of the lack of the api.
> 
> BUG=b/10538425
> TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release  -f *InitialPlayoutDelayTest*'
> R=henrika@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2144004

TBR=dwkang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2160006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-04 07:47:39 +00:00
dwkang@webrtc.org
310ac91d2a Enable SetInitialPlayoutDelay on Android. 
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.

BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release  -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2144004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4671 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-04 01:19:12 +00:00
tina.legrand@webrtc.org
ee92b664b3 Re-organizing ACM tests 
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret.

While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL.

I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD().

BUG=issue2173
R=minyue@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1961004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 07:33:51 +00:00
pbos@webrtc.org
2ab209ef14 Remove include_dirs from test/test.gyp. 
This is a cleanup step for having root-relative includes, include_dirs shouldn't be needed anymore.

BUG=1662
R=phoglund@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1984004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 08:49:48 +00:00
tina.legrand@webrtc.org
bd21fb5f8d Adding call to Opus PLC 
NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.

BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1727004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 11:01:07 +00:00
pwestin@webrtc.org
401ef361ac Added configuration of max delay to ACM and NetEq 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1964004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 21:01:36 +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