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17 Commits

Author SHA1 Message Date
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
henrik.lundin@webrtc.org
184b913eb5 Rename RTPanalyze to rtp_analyze and remove old version 
The tool RTPanalyze (used to process an input RTP dump into a
text file of RTP header info) was present in both the neteq and
neteq4 folders. This change pulls in changes from the old to the new
and renames the source file and tool to rtp_analyze.

Removing special code for dummy-rtp files (it is supported without
special code), and making the RED payload type settable using flags.

Moving from test/ to tools/ folder.

BUG=2692
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 20:56:17 +00:00
henrik.lundin@webrtc.org
b477fa6d21 Small fixes to plot_neteq_delay.m 
Fixing problems with wrap-arounds and other small things. Adding an
extra output value.

Review URL: https://webrtc-codereview.appspot.com/4929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 12:28:47 +00:00
henrik.lundin@webrtc.org
5ecdef11cc Do not use recursive calling in NetEq test tools 
This CL removes recursive calling in:
- NETEQTEST_DummyRTPpacket::readFromFile,
- NETEQTEST_RTPpacket::readFromFile.

The files currently exist for both NetEq3 and NetEq4, and all are
changed with this CL.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 08:26:49 +00:00
henrik.lundin@webrtc.org
5a43370cdb Dedicated speed test for NetEq3 
This is the same test as was aleready implemented for NetEq3 in r4763.

BUG=1363
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 20:58:33 +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
pbos@webrtc.org
2d1a55caed Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ 
BUG=163
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1900004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:54:00 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
pbos@webrtc.org
0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:28:06 +00:00
turaj@webrtc.org
6388c3e2fd Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. 
TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-12 21:42:18 +00:00
henrik.lundin@webrtc.org
75f8c78d08 Fixing path to ptypes.txt in NetEqRTPplay 
The default path to the file ptypes.txt needed by NetEqRTPplay
had gone old. Updating to new repo layout.

Also purging old payload types from the file itself.

Review URL: https://webrtc-codereview.appspot.com/966035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 15:22:00 +00:00
turaj@webrtc.org
b0dff12d2b 48 kHz extension to iSAC. 
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 17:43:52 +00:00
tina.legrand@webrtc.org
d0acdf6e2b Fix ordered comparison warnings in the RTPtimeshift unit test 
Original CL uploaded here: http://review.webrtc.org/933013/

Removing the checks if (argv[4/5] >= 0), they are not doing anything useful.

BUG=

Review URL: https://webrtc-codereview.appspot.com/933019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-16 15:34:32 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/ 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-22 18:19:23 +00:00