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218 Commits

Author SHA1 Message Date
mflodman@webrtc.org
90071dd647 Added API to set RTP timestamp offset extension. 
BUG=745

Review URL: https://webrtc-codereview.appspot.com/710011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 17:13:27 +00:00
braveyao@webrtc.org
52ce651283 Fix the auido noise issue with FEC enabled 
BUG = issue 652
TEST=manual test
Review URL: https://webrtc-codereview.appspot.com/720006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 07:30:08 +00:00
kma@webrtc.org
de91bf7cdc Changed compiler assert code so that it can be used by both C and C++ source files. 
Review URL: https://webrtc-codereview.appspot.com/733005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-11 00:13:25 +00:00
mflodman@webrtc.org
ba853c9970 Fixes for RTP extension time offset. 
Review URL: https://webrtc-codereview.appspot.com/720008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 14:30:53 +00:00
marpan@webrtc.org
4889120a84 Fix integer divisin truncation error. 
Patch fix from: thakis@chromium.org
https://webrtc-codereview.appspot.com/717006/

TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/721006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-03 17:33:13 +00:00
marpan@webrtc.org
8639fd9341 Use correct rtp header size for FEC packets. 
This is needed as of r2489 which introduced the header extension.

This also is a fix for issue 701.
Review URL: https://webrtc-codereview.appspot.com/708005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-30 18:17:02 +00:00
marpan@webrtc.org
71707aaae8 Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 
Review URL: https://webrtc-codereview.appspot.com/682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 16:27:51 +00:00
stefan@webrtc.org
ddfdfed3b5 Pass capture time (wallclock) to the RTP sender to compute transmission offset 
- Change how the transmission offset is calculated, to
  incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
  We must use the same clock as in the RTP module to be able to measure
  the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/666006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 13:21:22 +00:00
pwestin@webrtc.org
1853005f37 Change clock to be 64 bits in RTP module 
Review URL: https://webrtc-codereview.appspot.com/678011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 10:41:54 +00:00
tommi@webrtc.org
fb933bdb26 Landing: https://webrtc-codereview.appspot.com/680005/ 
Fix more -Wunused-private-field violations.
Review URL: https://webrtc-codereview.appspot.com/668010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 08:19:12 +00:00
asapersson@webrtc.org
d2e6779565 Fix for negative transmission time offset. 
Review URL: https://webrtc-codereview.appspot.com/671006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-28 07:53:15 +00:00
andrew@webrtc.org
81cf5e4752 Move test to src/test. 
- Refer to top-level directories by <(DEPTH), e.g. <(DEPTH)/testing.
- Remove now unneeded third_party_root.

TBR=henrike@webrtc.org
BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/669007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 01:41:54 +00:00
astor@webrtc.org
bd7aeba8fb Expose a set of options to the OveruseDetector supporting experiments 
Updated overuse_detector.* to use google style naming convention
Removed OveruseDetector::Reset
Review URL: https://webrtc-codereview.appspot.com/666005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-26 10:47:04 +00:00
henrike@webrtc.org
643be71700 Adds variable for third party directory. 
BUG=348
TEST=Manual testing in Chrome and WebRTC workspace.

Review URL: https://webrtc-codereview.appspot.com/674005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-25 10:48:58 +00:00
stefan@webrtc.org
190541578a Correct gypi files to match the actual filenames. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/656005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-18 10:57:05 +00:00
marpan@webrtc.org
c35f1d26c5 FEC: Fix to coverity issue 14448: unintended sign extension. 
Review URL: https://webrtc-codereview.appspot.com/647004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-12 20:12:13 +00:00
stefan@webrtc.org
9354cc965c Refactoring the receive-side bandwidth estimation into its own module. 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/637009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-07 08:10:14 +00:00
marpan@webrtc.org
5f97232cac Removing a TODO in the FEC: renaming the exisiting packets mask to indicate random mode, 
and refactored and renamed corresponding table file.
Review URL: https://webrtc-codereview.appspot.com/632007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-06 22:34:38 +00:00
marpan@webrtc.org
8866bb1132 FEC: Added another set of packet masks for the FEC. 
These FEC codes perform better for bursty (consecutive loss) 
than the existing set (which were designed for random loss). 
Updates to the unittests and test_fec accordingly.
Review URL: https://webrtc-codereview.appspot.com/581005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 16:42:20 +00:00
bjornv@webrtc.org
cb89c6f914 Revert 2363 - Refactoring the receive-side bandwidth estimation into its own module. 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/605007

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/634006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 12:25:35 +00:00
stefan@webrtc.org
f72881406f Refactoring the receive-side bandwidth estimation into its own module. 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/605007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-05 10:44:00 +00:00
marpan@webrtc.org
899baa821b Temporarily disable first partition packet counting to avoid a bug in ProducerFec which doesn't properly handle important packets. 
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/631005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 18:32:16 +00:00
tina.legrand@webrtc.org
4517585db5 Adding separate payload types for stereo modes 
BUG=Issue 452
TEST=audio_coding_test, voe_auto_test, voe_cmd_test

Edit: adding Patrik to review:
src/modules/rtp_rtcp/source/rtp_receiver.cc
...and Shijing to review:
src/voice_engine/main/source/channel.cc
src/voice_engine/main/test/cmd_test/voe_cmd_test.cc

Review URL: https://webrtc-codereview.appspot.com/540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 09:27:35 +00:00
pwestin@webrtc.org
c2722a0e68 Fixed compiler warning 
Review URL: https://webrtc-codereview.appspot.com/624005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-01 08:56:42 +00:00
phoglund@webrtc.org
baaf2434a7 Extracted a method for sending padded data. 
BUG=
TEST=Ran vie_auto_test and voe_auto_test standard tests.

Review URL: https://webrtc-codereview.appspot.com/605004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-31 10:47:35 +00:00
pwestin@webrtc.org
1eef9c16ff Bitrate bugfixes 
Review URL: https://webrtc-codereview.appspot.com/609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-29 09:28:43 +00:00
marpan@webrtc.org
747cd87da1 FEC: For multi-frame FEC, allow for the size of the code to be increased, 
under certain conditions. This generally improves the FEC recovery for 
     bursty loss under medium-high protection level.
Review URL: https://webrtc-codereview.appspot.com/566012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 16:50:00 +00:00
astor@webrtc.org
e231c61ba7 Style guide reformatting of overuse_detector.{h,cc} 
Review URL: https://webrtc-codereview.appspot.com/575006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-22 12:24:00 +00:00
pwestin@webrtc.org
b2179c20f0 Bugfix issue 533. Client does not handle NACK or PLI requests received from far end if a RTCP report from it has not been processed when RFC 5506 is enabled. 
Review URL: https://webrtc-codereview.appspot.com/569020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-21 12:00:49 +00:00
stefan@webrtc.org
1bca6d2437 Fixes coverity warnings. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/566014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-21 07:42:12 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module. 
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 11:08:54 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 23:01:04 +00:00
turaj@webrtc.org
980d6be535 Revert 2212 - Bug fix 
Review URL: https://webrtc-codereview.appspot.com/576009

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/575009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 22:57:21 +00:00
turaj@webrtc.org
449b525453 Revert 2214 - Bugfix 
A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR
Review URL: https://webrtc-codereview.appspot.com/570007

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/571010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 22:54:39 +00:00
pwestin@webrtc.org
fd3fef514b Bugfix 
TBR
Review URL: https://webrtc-codereview.appspot.com/570007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 15:14:41 +00:00
pwestin@webrtc.org
e1c97a2723 Bug fix 
Review URL: https://webrtc-codereview.appspot.com/576009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 13:55:23 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 12:33:50 +00:00
hta@webrtc.org
b6f2417f37 Renamed all _test.cc files to _unittest.cc, to conform to convention 
for webrtc.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/560004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-04 08:13:57 +00:00
hta@webrtc.org
54536bb6d4 Refactoring of the TMMBRSet class, giving it a reasonably tight interface. 
The CL also fixes a number of line length and tab issues in touched files.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/553005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-03 14:07:23 +00:00
hta@webrtc.org
404843e6e5 Timeout tests for TMMBR 
Added a test that injects 3 packets and then times out.

Rewrote the packet construction in test to use a builder format.
This makes tests a lot more readable.

Odd behaviour of timeout found; documented as test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/553004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 09:56:45 +00:00
hta@webrtc.org
3c0df7d376 Fixed a build break: I'd forgotten to remove a DELETE statement. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/555004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 08:22:21 +00:00
hta@webrtc.org
47059b5dfb Adds unit tests for RTCP receiver, focusing on TMMBR handling. 
This is the first part of a plan:

- Get basic unit tests for rtcp_receiver.
- Get an unit test for some code inside rtcp_receiver
  that touches the TMMBRSet class in hard-to-decipher ways
  (rtcp_receiver_help, GetTMMBRSet function, use of memmove()).
- Refactor the TMMBRSet class.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/547005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 07:46:22 +00:00
hta@webrtc.org
65a4e4ed56 Minor refactoring: RTCPReceiver::BoundingSet 
Remove ability to modify a pointer argument.

Added a test for transmitting a non-empty TMMBN

BUG=
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/542004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 11:23:41 +00:00
hta@webrtc.org
c2d985257b untabify 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 08:25:10 +00:00
hta@webrtc.org
9d54cd12ab TMMBN sender test passes, fixed receiver flag bug 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 08:24:55 +00:00
pwestin@webrtc.org
b1313aac7c Fix missing h file change. 
Review URL: https://webrtc-codereview.appspot.com/535001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:42:05 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
tommi@webrtc.org
a990e122c4 * Change the reference counting implementation for VoE to be per object and 
not per interface. This simplifies things a bit, reduces code and makes it
  possible to implement reference counting (if we ever want) without the
  static Delete() method.  (Reference counted objects are traditionally
  implicitly deleted via the last Release())

* Since the reference counting code is now simpler, there's no need for the
  RefCount class so I'm removing it.

* Also removing the optional |ignoreRefCounters| variable from the VoiceEngine::Delete
  method.  The justification is that it's no longer used and the reason it was there
  in the first place was to avoid bugs in third party code, so it's an indication that
  something is wrong elsewhere.

* Fix a crash in voe_network_impl and voe_extended_test that would happen on machines with IPv6 support.

* Fix an assert (handle the case) in the extended audio tests when SetCNAME is called with a NULL pointer.

* As a side effect of this change, hopefully the footprint of VoE will be slightly smaller :)

BUG=10445 (Coverity)
Review URL: https://webrtc-codereview.appspot.com/521003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 15:28:22 +00:00
asapersson@webrtc.org
d18dd6dc7e Made OnPacketLossStatisticsUpdate function virtual (needed for ViCE). 
Review URL: https://webrtc-codereview.appspot.com/520002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-25 07:19:02 +00:00
pwestin@webrtc.org
ddab60be56 Wire up pading. 
Review URL: https://webrtc-codereview.appspot.com/509002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 14:52:15 +00:00