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Author SHA1 Message Date
Peter Kasting
83ad33a8ae Upconvert various types to int. 
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.

Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."

This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t).  Other locations will be converted to size_t in a separate change.

BUG=none
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54629004

Cr-Commit-Position: refs/heads/master@{#9405}
 2015-06-10 00:20:09 +00:00
Peter Kasting
248b0b0790 Run clang-format --style=Chromium on four files I'm otherwise touching. 
The existing style in these files is pretty inconsistent and wildly divergent
from most of WebRTC/Chromium; clang-formatting them not only makes them easier
to read, it makes me see fewer presubmit errors when I try to touch the files to
make other changes.

BUG=none
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52019004

Cr-Commit-Position: refs/heads/master@{#9364}
 2015-06-03 19:32:55 +00:00
Minyue Li
c065cc797d Clarify boolean flags in neteq_opus_quality_test. 
Note that the use of boolean flags in gflags is a bit unnatural. For setting a boolean flag to false: putting "no" in front of its name (see http://gflags.github.io/gflags/)

We make this clearer by defaulting boolean flags to false, and clarifying it in the description.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53539004

Cr-Commit-Position: refs/heads/master@{#9293}
 2015-05-27 08:01:18 +00:00
Bjorn Volcker
de4703c5d1 Refactor common_audio/vad: Create now returns the handle directly instead of an error code 
Changed the WebRtcVad_Create() function to the more conventional format of returning the handle directly instead of an error code to take care of.
In addition NULL was changed to nullptr in the files where it applied.

Affected components:
* AGC
* VAD
* NetEQ

BUG=441, 3347
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51919004

Cr-Commit-Position: refs/heads/master@{#9291}
 2015-05-27 05:23:11 +00:00
Minyue Li
f761d10393 Update NetEq Quality Test. 
1. move channel number of input file to the base class

2. limit channel number to be 1, since the resampler support only mono at the moment

3. adding a logging function

4. adding more switch to neteq_opus_quality_test

BUG=2692
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47239004

Cr-Commit-Position: refs/heads/master@{#9260}
 2015-05-22 09:21:58 +00:00
Henrik Lundin
8171735b0c Add NetEqIlbcQualityTest 
This is virtually the same as NetEq{Isac,Opus}QualityTest but for iLBC.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50909004

Cr-Commit-Position: refs/heads/master@{#9178}
 2015-05-12 13:04:29 +00:00
Henrik Lundin
e5ff00a1c6 Add NetEqPcmuQualityTest 
This is virtually the same as NetEq{Isac,Opus}QualityTest but for PCMu.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54379004

Cr-Commit-Position: refs/heads/master@{#9176}
 2015-05-12 10:09:53 +00:00
Henrik Lundin
83b5c053b9 Modify NetEqQualityTest 
- Take input sample rate as parameter - provides resampling when needed.
- Add support for wav output.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49699004

Cr-Commit-Position: refs/heads/master@{#9158}
 2015-05-08 08:34:00 +00:00
Ivo Creusen
5a3178042b Reformatting RTPtimeshift.cc file. 
BUG=2692
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45239004

Cr-Commit-Position: refs/heads/master@{#9052}
 2015-04-22 11:11:39 +00:00
minyue@webrtc.org
8f76cd25ec Renaming neteq_opus_fec_quality_test. 
neteq_opus_fec_quality_test has been modified to test more configurations of Opus than only FEC. It makes sense to rename it to neteq_opus_quality_test. This was planned in

https://webrtc-codereview.appspot.com/45619004/

but was forgotten. This CL handles it, and makes it easy for review.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45709004

Cr-Commit-Position: refs/heads/master@{#8782}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 20:44:26 +00:00
minyue@webrtc.org
7f7d7e3427 Prevent crash in NetEQ when decoder overflow. 
NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.

The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.

BUG=4361
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45619004

Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 12:31:19 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 13:04:54 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
pkasting@chromium.org
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : 
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 21:29:45 +00:00
kwiberg@webrtc.org
648f5d6dc7 pcm16b: Make input arrays const and use uint8_t[] for byte arrays 
There were both uint8 and uint16 versions of the pcm16b encode and
decode functions; this patch removes the latter.

BUG=909
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34139004

Cr-Commit-Position: refs/heads/master@{#8309}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-10 09:19:09 +00:00
kwiberg@webrtc.org
1c6239a3b6 G711: Make input arrays const and use uint8_t[] for byte arrays 
BUG=909
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39809004

Cr-Commit-Position: refs/heads/master@{#8294}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 12:56:16 +00:00
minyue@webrtc.org
7dba7860c7 Setting Opus target application. 
This CL is to allow to set Opus target application at the creation of an encoder.

According to Opus spec, there are three applications:

OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY

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

Review URL: https://webrtc-codereview.appspot.com/37479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 16:01:50 +00:00
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory. 
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:30:52 +00:00
pbos@webrtc.org
e728ee03ba Remove or rename typedefs with _t prefixes. 
_t prefixes are reserved for additional typenames in POSIX.

R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/36559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 13:43:55 +00:00
minyue@webrtc.org
0ca768b131 Adding DTX to WebRTC Opus wrapper (relanding). 
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.

See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/

Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 16:09:35 +00:00
pbos@webrtc.org
d8ca723de7 Remove CELT support from audio_coding. 
R=henrik.lundin@webrtc.org, juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 11:49:13 +00:00
minyue@webrtc.org
19dd129c69 Revert 7846 "Adding DTX to WebRTC Opus wrapper" 
> Adding DTX to WebRTC Opus wrapper
> 
> This is a step toward adding Opus DTX support in WebRTC.
> 
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
> 
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
> 
> We transmit the first 1-byte packet to let decoder be in-sync
> 
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13219004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 15:11:15 +00:00
minyue@webrtc.org
4321f175f1 Adding DTX to WebRTC Opus wrapper 
This is a step toward adding Opus DTX support in WebRTC.

Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See

https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html

We transmit the first 1-byte packet to let decoder be in-sync

BUG=webrtc:1014
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 13:27:39 +00:00
kwiberg@webrtc.org
e04a93bcf5 Move the AudioDecoder interface out of NetEq 
It belongs with the codecs, next to the AudioEncoder interface.

R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org

Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:12:53 +00:00
kwiberg@webrtc.org
cb858ba397 Make an AudioEncoder subclass for iLBC 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@google.com
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 17:11:44 +00:00
kwiberg@webrtc.org
3800e13a3a Revert r7798 ("Move the AudioDecoder interface out of NetEq") 
Apparently, it caused all sorts of problems I don't have time to
straighten out right now.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 16:28:17 +00:00
kwiberg@webrtc.org
00ba1a7dfd Move the AudioDecoder interface out of NetEq 
It belongs with the codecs, next to the AudioEncoder interface.

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 14:23:23 +00:00
kwiberg@webrtc.org
0cd5558f2b AudioEncoder subclass for G722 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 11:45:51 +00:00
henrik.lundin@webrtc.org
20446e7e56 Move and rename neteq/test/RTPcat to neteq/tools/rtpcat 
BUG=2692
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 14:23:01 +00:00
henrik.lundin@webrtc.org
83317146ba Adding a new test helper RtpFileWriter and use it in RTPcat 
This new helper class writes RTP packets to file in rtpdump format.
A unit test is also included.

The new test class is used while re-writing the test tool RTPcat.

BUG=2692
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 11:25:04 +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
kwiberg@webrtc.org
c78cf97ecb Remove the useless dummy state parameter to WebRtcG711_* 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:23:36 +00:00
kwiberg@webrtc.org
1172988c79 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[] 
The affected functions are

  WebRtcIsacfix_ReadFrameLen
  WebRtcIsacfix_GetNewBitStream
  WebRtcIsacfix_ReadBwIndex

and

  WebRtcIsac_ReadFrameLen
  WebRtcIsac_GetNewBitStream
  WebRtcIsac_ReadBwIndex
  WebRtcIsac_GetRedPayload

BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 10:53:42 +00:00
kwiberg@webrtc.org
7ee24a7906 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t 
We have to fix both at once, since there's a macro that calls one of
them or the other.

BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7266

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 10:31:02 +00:00
andrew@webrtc.org
a3c4d4dd2c Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." 
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795

> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
> 
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
> 
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19229004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 01:32:57 +00:00
kwiberg@webrtc.org
8c5740b485 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t 
We have to fix both at once, since there's a macro that calls one of
them or the other.

BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 23:04:14 +00:00
andresp@webrtc.org
76ba7caae8 Re-enable neteq_performance_unittest.cc for android. 
BUG=3770
R=kjellander@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 12:29:50 +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
kjellander@webrtc.org
6f729e8a74 Disable video_engine_tests and webrtc_perf_tests on Android. 
BUG=3770
TESTED=Running the tests locally on an Android device.
R=phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-02 15:13:55 +00:00
fbarchard@google.com
d7b4dea801 initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 
BUG=3660
TESTED=set DEPOT_TOOLS_WIN_TOOLCHAIN=0 & set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 &ninja -C out\Debug
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:46:42 +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
6568e97d10 This is to compare NetEq with various codecs under a shared packet loss pattern. 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-25 12:17:41 +00:00
henrik.lundin@webrtc.org
d8de0669c9 Revert "Restore ptypes.txt file" 
This reverts r6460. It turns out the file was no longer needed after
all.

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-18 11:09:53 +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
henrik.lundin@webrtc.org
2ca2188906 Restore ptypes.txt file 
The file was lost when the neteq folders where moved and renamed.

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 08:51:01 +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
henrik.lundin@webrtc.org
9c55f0f957 Rename neteq4 folder to neteq 
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.

This CL effectively reverts r6257 "Rename neteq4 folder to neteq".

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-09 08:10:28 +00:00
henrik.lundin@webrtc.org
1b9df05c85 Revert 6257 "Rename neteq4 folder to neteq" 
> Rename neteq4 folder to neteq
> 
> BUG=2996
> R=turaj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12569005

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 07:33:39 +00:00
henrik.lundin@webrtc.org
a90f6d67f7 Rename neteq4 folder to neteq 
BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 06:23:34 +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