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Author SHA1 Message Date
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
turaj@webrtc.org
0fc2558503 Add turaj@webrtc.org to NetEq owners. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1980004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 17:07:18 +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
henrike@webrtc.org
89c674053e Adds all unittests to android NDK-APK framework. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 16:53:47 +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
turaj@webrtc.org
fee739c224 Risk of division by zero. 
bug=b9338699

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1634004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 20:10:06 +00:00
turaj@webrtc.org
a305e9612a Nack for audio. 
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1507004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 19:00:09 +00:00
turaj@webrtc.org
3942f3a985 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. 
bug=issue1847

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 21:31:22 +00:00
kjellander@webrtc.org
5156c94f89 Disable neteq_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1460
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1595004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:47:24 +00:00
turaj@webrtc.org
e46c8d3875 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. 
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:39:43 +00:00
turaj@webrtc.org
8630cfe016 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. 
BUG=issue1770
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 23:54:54 +00:00
sergeyu@chromium.org
bd4a2feddb Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). 
BUG=1725
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1395004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 18:11:36 +00:00
andrew@webrtc.org
342353780d Consolidate common_audio into a single target. 
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.

R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:43:26 +00:00
turaj@webrtc.org
28d54ab18f Improve AV-sync when initial delay is set and NetEq has long buffer. 
Review URL: https://webrtc-codereview.appspot.com/1324006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 18:53:35 +00:00
turaj@webrtc.org
92d1f07551 Elevate NetEq short-term activity statistics to ACM level for logging. 
Review URL: https://webrtc-codereview.appspot.com/1313004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 16:52:04 +00:00
pbos@webrtc.org
e4b6064f8e Replace legacy G_CONST with const. 
BUG=1608

Review URL: https://webrtc-codereview.appspot.com/1310005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 18:06:57 +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
kjellander@webrtc.org
fa53d8717c Fixing/disabling Windows x64 warnings 
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.

With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.

BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64

Review URL: https://webrtc-codereview.appspot.com/1060008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-04 10:07:17 +00:00
turaj@webrtc.org
763faeab4e Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. 
This is discovered during a test for controlling delay. It is not simple to reproduce it. 

Bug=
test=manual test verified that |functionDurationEst| is correctly updated.
Review URL: https://webrtc-codereview.appspot.com/1074013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 18:21:58 +00:00
henrik.lundin@webrtc.org
73deaadd0e Removing a hack for CNG 
However, two other "hacks" had to be added to maintain bit-exactness
with legacy.

Note that this change requires a new version of the universal.rtp test
input, although the output reference stays the same.

Moving reference files, and using a new input vector for NetEq4.
The new input vector neteq_universal_new.rtp is identical to the old
neteq_universal.rtp, except that the payload type for CNG packets that
follows a wideband codec is changed to 98.

Update to resources revision 15 where the new reference files are.

Also changing a faulty log error.

Review URL: https://webrtc-codereview.appspot.com/1078009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-31 13:32:51 +00:00
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
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
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
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
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
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
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
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
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
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
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
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