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147 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
d94659dc27 Initial upload of NetEq4 
This is the first public upload of the new NetEq, version 4.

It has been through extensive internal review during the course of
the project.

TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1073005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-29 12:09:21 +00:00
kma@webrtc.org
c4373bc737 Moved several function pointer declarations in iSAC to isac initialization file. 
Fixed clang linker problem of not being able to find symbols.
Review URL: https://webrtc-codereview.appspot.com/1061006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 04:55:21 +00:00
kma@webrtc.org
16d540eff1 Fixed text relocation code related to ARM assembly code. 
Refer to WebRTC issue 1300.
Review URL: https://webrtc-codereview.appspot.com/1055004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 03:18:05 +00:00
kma@webrtc.org
e8482f0e9f Revert 3406 
> Moved all function pointer declarations in iSAC to a single place.
> Review URL: https://webrtc-codereview.appspot.com/1057006

TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1074005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 23:57:56 +00:00
niklas.enbom@webrtc.org
cd2f1356ee Revert 3405 
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1074004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 22:05:30 +00:00
kma@webrtc.org
ebef7e4ac1 Moved all function pointer declarations in iSAC to a single place. 
Review URL: https://webrtc-codereview.appspot.com/1057006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 21:19:24 +00:00
niklas.enbom@webrtc.org
05e7bfeeea Mainly hlundin's patch. 
Review URL: https://webrtc-codereview.appspot.com/1052004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 18:53:43 +00:00
kma@webrtc.org
4782911572 Optimized WebRtcIsacfix_Time2Spec() for iSAC-Fix in ARM Neon processor. 
Review URL: https://webrtc-codereview.appspot.com/1005004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 01:37:33 +00:00
henrik.lundin@webrtc.org
5dfb1f2cd3 Bug fix in WebRtcOpus_DurationEst 
The function WebRtcOpus_DurationEst returned the number of samples
per packet in the native 48 kHz sample rate, while the decoder
function returns data in 32 kHz sample rate. This creates a discrepancy
that makes NetEQ's lip-sync functionality add too little delay.

BUG=1334
TEST=try bots

Review URL: https://webrtc-codereview.appspot.com/1069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-23 11:57:03 +00:00
henrike@webrtc.org
09738616de Fixes payload spelling error. 
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1052006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 16:43:45 +00:00
wjia@webrtc.org
a3c82bf667 Remove '<(library)' in gyp files. 
This will remove all usage of '<(library)' in all webrtc gyp files. 
Review URL: https://webrtc-codereview.appspot.com/1049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 23:42:21 +00:00
tina.legrand@webrtc.org
d0d41498a3 Adding AUDIO application as default for Opus stereo 
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.

I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.

Next step is to add an API to choose application mode.

BUG=issue1239

Review URL: https://webrtc-codereview.appspot.com/1007006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:23:10 +00:00
leozwang@webrtc.org
ac77084583 Roll opus to 172355 and delete opus_demo from webrtc opus 
opus_demo has been inlucded in opus in chromium.

BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/973013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 17:24:30 +00:00
tina.legrand@webrtc.org
4275ab1ca0 Implement NetEq duration estimation for Opus. 
Review URL: https://webrtc-codereview.appspot.com/983004
Patch from Ralph Giles <giles@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 09:52:45 +00:00
fbarchard@google.com
3c37354b70 Initialize 3 variables which are preventing VS2012 from building. 
BUG=1211
TESTED=ninja -C out\Release
Review URL: https://webrtc-codereview.appspot.com/992005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-15 01:09:18 +00:00
roosa@google.com
b8ba4d8109 Add number of inserted samples to NetEq statistics. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/964030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 00:06:18 +00:00
turaj@webrtc.org
c454fab03b Reformatting ACM. All changes are bit-exact in this CL. 
TEST=VoE auto-test, audio_coding_module_test; 

only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision
Review URL: https://webrtc-codereview.appspot.com/937035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 22:46:43 +00:00
fbarchard@google.com
86464eacb6 ISAC_main_inst initialized to NULL to avoid potentially garbage pointer passed to WebRtcIsacfix_EncoderInit 
BUG=1211
TESTED=local build on Windows.  Failed previously with vs2012.  With this change kenny.cc builds.
Review URL: https://webrtc-codereview.appspot.com/984004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 07:47:54 +00:00
kma@webrtc.org
fa5b6bf4f4 Optimized WebRtcIsacfix_Spec2Time() for iSAC-Fix in ARM Neon processor. Speed doubled. 
Review URL: https://webrtc-codereview.appspot.com/930033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:00:52 +00:00
roosa@google.com
b718619f0a Expose NetEq playout mode off through VoiceEngine. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/971016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 21:59:14 +00:00
turaj@webrtc.org
36965b1803 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. 
TBR=tlegrand@google.com

TEST=voe_cmd_test, ACM unittest.
Review URL: https://webrtc-codereview.appspot.com/974011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 23:52:43 +00:00
kma@webrtc.org
0e739508e0 Added buildbot benchmarking in iSAC and APM into Android platform build. 
Review URL: https://webrtc-codereview.appspot.com/964022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 15:26:28 +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
226db898f7 Dual-stream implementation, not including VoE APIs. 
Review URL: https://webrtc-codereview.appspot.com/933015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 22:13:31 +00:00
turaj@webrtc.org
277ec8e3f5 Fix a bug when iSAC-48kHz was added. 
I discovered this by running extended VoE test on "Codecs."

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/973010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 21:16:23 +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
5b4fe494e7 Changing default bitrate to 64000 bps for Opus. 
Default settings for Opus in WebRtc is stereo, but we had default rate to 32 kbps. This is too low for stereo, where we need to encode using 64 kbps to get the quality we like.

BUG=

Review URL: https://webrtc-codereview.appspot.com/974008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 12:08:53 +00:00
kjellander@webrtc.org
ad0f3baf90 Removing redundant codec unittest targets. 
The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests

Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).

The following test has been removed since it was empty:
* audio_conference_mixer_unittests

BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)

Review URL: https://webrtc-codereview.appspot.com/971008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 10:52:29 +00:00
henrik.lundin@webrtc.org
8552c71290 Fixing neteq_unittests for VS 2012 
For Visual Studio versions older than 2012, we are using a
separate reference output file for windows. (All other platforms
share the same generic reference file.) In VS 2012, the output
matches the generic reference, and not the platform-specific one.

Since, the ResourcePath() method cannot change behavior depending
on compiler version, this fix will short-cut ResourcePath() for
VS 2012 or newer (_MSC_VER >= 1700).

Also made NetEqDecodingTest.TestBitExactnes stop on the first diff.
Once there is a difference, the output is no longer bit-exact, and
the test should be declared a failure.

BUG=
TEST=neteq_unittests on VS2012, try bots

Review URL: https://webrtc-codereview.appspot.com/966028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-29 12:03:18 +00:00
henrik.lundin@webrtc.org
f826bb6fb2 Fixing a bug related to RCU in NetEQ 
RCU was disabled due to that the RCU flag was overwritten with zero
in the packet buffer.

BUG=1156
TEST=trybots, neteq_unittests, audio_coding_module_test, audio_coding_unittests

Review URL: https://webrtc-codereview.appspot.com/969012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-29 07:32:38 +00:00
tina.legrand@webrtc.org
c4590580e8 Opus mono/stereo on the same payloadtype, and fix of memory bug 
During call setup Opus should always be signaled as a 48000 Hz stereo codec, not depending on what we plan to send, or how we plan to decode received packets.
The previous implementation had different payload types for mono and stereo, which breaks the proposed standard.

While working on this CL I ran in to the problem reported earlier, that we could get a crash related to deleting decoder memory. This should now be solved in Patch Set 3.

BUG=issue1013, issue1112

Review URL: https://webrtc-codereview.appspot.com/933022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-28 12:23:29 +00:00
andrew@webrtc.org
b43502e388 Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots. 
Review URL: https://webrtc-codereview.appspot.com/929022

TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 23:57:38 +00:00
kma@webrtc.org
4cd8f1f182 Added performance benchmarking in APM and iSAC-fix for Buildbots. 
Review URL: https://webrtc-codereview.appspot.com/929022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 22:02:47 +00:00
henrika@webrtc.org
5ba3decc94 Ensures that we can build using VS 2012 on Windows. 
See more details at https://code.google.com/p/webrtc/issues/detail?id=1146&

TBR=Niklas
BUG=1146

Review URL: https://webrtc-codereview.appspot.com/939028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 09:12:02 +00:00
andrew@webrtc.org
4b97793f91 Ensure opus_demo has a targets block. 
TBR=leozwang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-21 20:16:53 +00:00
leozwang@webrtc.org
cfcadab8d4 Build opus_demo 
BUG=1082
TEST=trybots

With this cl we can build opus_demo.
Review URL: https://webrtc-codereview.appspot.com/936029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-21 18:13:46 +00:00
tina.legrand@webrtc.org
5ac387c4d1 Allow NetEQ to use real packet durations. 
This is a copy of http://review.webrtc.org/864014/

This adds a FuncDurationEst to each codec instance which estimates
the duration of a packet given the packet contents and the duration
of the previous packet. By default, this simply returns the
duration of the previous packet (which is what is currently assumed
to be the duration of all future packets). This patch also provides
an initial implementation of this function for G.711 which returns
the actual number of samples in the packet.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/935016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-19 08:02:55 +00:00
kma@webrtc.org
55cd78cfc2 Porting ARM optimization from Android to ios. 
Tested APM and iSAC in Android. Bit-exact with original versions.
Changes include removing or changing some GCC derivatives (e.g. .fnstart, .hword), instruction syntax, etc.
Review URL: https://webrtc-codereview.appspot.com/934009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-17 00:22:46 +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
kma@webrtc.org
1786436eb2 Pure Neon assembly coding for WebRtcIsacfix_AutocorrNeon() in iSAC-Fix. 
Review URL: https://webrtc-codereview.appspot.com/939018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 18:44:24 +00:00
tina.legrand@webrtc.org
7577ddf27b Refactoring acm_generic_codec 
First patch: updating comments.

BUG=1024

Review URL: https://webrtc-codereview.appspot.com/936019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 14:09:35 +00:00
kma@webrtc.org
fa65c851fe Optimized function AllpassFilter2FixDec16() in isac fix for Android Neon platforms. 
With an offline test, codec cycles were reduced by 4%.
Review URL: https://webrtc-codereview.appspot.com/936007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 00:39:45 +00:00
tina.legrand@webrtc.org
0ad3c1af0a Adding Opus stereo support to WebRTC 
This CL adds support for sending and receiving stereo using the Opus codec.

BUG=issue1013

Review URL: https://webrtc-codereview.appspot.com/930008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 08:07:29 +00:00
tina.legrand@webrtc.org
f7fa6276e2 Reformating files in audio coding module. 
This CL format the ramaining files on the audio coding module. No other changes are done, except for fixing a few long lines and TODOs without owner.

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/928012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-05 09:35:51 +00:00
andrew@webrtc.org
a3ae6dc846 Remove opus include path from opus.gypi. 
It's now exported by opus.gyp.

Review URL: https://webrtc-codereview.appspot.com/928007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 17:51:09 +00:00
tina.legrand@webrtc.org
1f8c02a4de Getting Opus from Chrome 
This CL replaces http://review.webrtc.org/921006/

BUG=issue982

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/933006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 12:37:08 +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