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Author SHA1 Message Date
mflodman@webrtc.org
b1fbf016b5 Added timestamp logs, i.e. only tracing. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/482001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-16 12:58:49 +00:00
tina.legrand@webrtc.org
16b6b90a82 Split of stereo packets moved 
In this CL I have rewritten the way we handle stereo packets in VoE.
Before this change we split the packets in the RTP module and added two packets to ACM, one for the left channel and one for the right. This lead to timing problems caused when a different thread called RecOut in between the two calls to add stereo packet to ACM. (RecOut is called to pull audio data, decode packets, on the receiving side).

While doing the change I also took the opportunity to changed some functions so that the data stream is uint8 everywhere.

The list of files in this CL is long, but should be fairly easy to review. It is difficult to see what has been changed  in some of the tests, but I can explain offline.

Reviewers:
Björn - /src/modules/audio_coding
Patrik - /src/modules/rtp_rtcp
Patrik -/src/modules/utility
Henrik A - /src/voice_engine

BUG=410
TEST=voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/473003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-12 11:02:38 +00:00
stefan@webrtc.org
c35f5ced92 Enable multi-frame FEC by default for temporal layers <= 2. For two temporal layers we currently only protect the base layer. 
This also introduces zero column insertion into packet masks when some sequence numbers deliberately haven't been given to the FEC generator.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/477001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-11 07:42:25 +00:00
tommi@webrtc.org
98ad0ff1b0 Move the COMPILE_ASSERT macro to its own header file. 
TEST=n/a
BUG=none
Review URL: https://webrtc-codereview.appspot.com/492002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-10 11:53:07 +00:00
pwestin@webrtc.org
cac787842c New attempt to cleanup TMMBR. 
Review URL: https://webrtc-codereview.appspot.com/472007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-05 08:30:10 +00:00
henrike@webrtc.org
0ad51862dc Revert 1961 - Clean up TMMBR handling. 
Review URL: https://webrtc-codereview.appspot.com/465001

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/473001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 16:54:13 +00:00
marpan@webrtc.org
3a6080d4c0 FEC Decoding: Insert the received media packets directly into VCM without going 
through the recovered packet list. Avoids an issue with very old re-transmitted packets.

Updated the receiver_fec unittest accordingly.
Review URL: https://webrtc-codereview.appspot.com/465003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 16:16:21 +00:00
pwestin@webrtc.org
20f4440c73 Clean up TMMBR handling. 
Review URL: https://webrtc-codereview.appspot.com/465001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 11:40:15 +00:00
mflodman@webrtc.org
534a435751 Removed RTP Keepalive from RTP module. 
Review URL: https://webrtc-codereview.appspot.com/455007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-27 06:57:41 +00:00
stefan@webrtc.org
38f247d800 Fixes an issue in the FEC decoder where a big jump in sequence numbers may cause new packets to be discarded if there is a wrap around. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/455003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-23 22:34:14 +00:00
stefan@webrtc.org
af5ffd5bb9 Fixes for coverity warnings. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/461001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-23 16:01:15 +00:00
tina.legrand@webrtc.org
c231e4cb03 Fixing bug in re-packing of stereo packets. 
BUG=341
TEST=voe_cmd_test, run G.722. First modify /src/modules/audio_coding_main/source/acm_codec_database.cc
@@ -149,7 +149,7 @@ const CodecInst ACMCodecDB::database_[] = {
   {kDynamicPayloadtypes[count_database++], "CELT", 32000, 320, 2, 64000},
 #endif
 #ifdef WEBRTC_CODEC_G722
-  {9, "G722", 16000, 320, 1, 64000},
+  {9, "G722", 16000, 320, 2, 64000},
 #endif
 #ifdef WEBRTC_CODEC_G722_1

Review URL: https://webrtc-codereview.appspot.com/454001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1930 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-23 10:01:11 +00:00
stefan@webrtc.org
e0d6fa4c66 Adding classes for handling multi-frame FEC. 
The FEC behavior is unchanged with this commit, we will still be
limited to FEC over one frame for now.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/450006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-20 22:10:56 +00:00
leozwang@webrtc.org
0975d2158c Cleanup messy data type of unknown_payload_type 
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 20:59:13 +00:00
mflodman@webrtc.org
9ec883e8bd Allow multiple REMB groups and introduce receive channels. 
BUG=312
TEST=ViE standard autotest and API test.

Review URL: https://webrtc-codereview.appspot.com/432005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 17:12:41 +00:00
mflodman@webrtc.org
fa6bc673b0 Changed default module condition for BW estimate. 
Review URL: https://webrtc-codereview.appspot.com/433001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 11:59:55 +00:00
leozwang@webrtc.org
42e362eee5 Fix compilation error on android 
Review URL: https://webrtc-codereview.appspot.com/426006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-02 17:14:09 +00:00
leozwang@webrtc.org
07c68b9c9d Correct wrong usage of WebRtc_Word8 in rtp and udp module 
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/418001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 16:09:51 +00:00
marpan@webrtc.org
946601e408 Change default packetization mode to an equal size mode. 
This will produce equal size packets for each frame, which should be somewhat more favorable (less overhead/padding data) for the FEC.
Review URL: https://webrtc-codereview.appspot.com/396013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 18:52:53 +00:00
niklas.enbom@webrtc.org
7cb0c240cb Trying to free up hellner from review work, since he mainly works in libJingle. 
Review URL: https://webrtc-codereview.appspot.com/392020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 13:58:58 +00:00
mflodman@webrtc.org
4cb060127c Disabled RTPModule VP8 packetizer assert. 
BUG=293

Review URL: https://webrtc-codereview.appspot.com/399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 13:07:01 +00:00
mflodman@webrtc.org
f7b6078f6f Allow multiple send channels for REMB. Current implementation splits the remote estimate evenly between all senders. 
This CL will be followed by a CL adding support for several REMB groups.

TEST=video_engine_core_unittests

Review URL: https://webrtc-codereview.appspot.com/394002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 14:50:24 +00:00
stefan@webrtc.org
439be29445 Add APIs for getting receive-side estimated bandwidth and codec target rate. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 14:45:37 +00:00
henrike@webrtc.org
d6d014ff12 Fixes memory leaks introduced in 1698. 
Review URL: https://webrtc-codereview.appspot.com/387014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 02:18:09 +00:00
henrike@webrtc.org
f5da4da409 Removes a global non POD instance from the RTP_RTCP module that was introduced in https://code.google.com/p/webrtc/source/detail?r=1076. 
Review URL: https://webrtc-codereview.appspot.com/314001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-15 23:54:59 +00:00
mflodman@webrtc.org
d7d46887a6 Update receive only channels with RTT. 
vie_autotest_loopback.cc will not be part of the commit, it's only to show the test.

TEST=temporary with attached loopback test.

Review URL: https://webrtc-codereview.appspot.com/390007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 12:49:59 +00:00
marpan@webrtc.org
567d507707 Fixes a bug when number of media packets in a frame is larger than maximum allowed for the generateFEC. 
Review URL: https://webrtc-codereview.appspot.com/391003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 18:56:14 +00:00
mflodman@webrtc.org
8224e19dd9 Fixed incorrect packet loss reported to encoder. 
BUG=275

Review URL: https://webrtc-codereview.appspot.com/394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:41:57 +00:00
pwestin@webrtc.org
5e954814a8 Clanup handling of key frame requests and FIR. 
Review URL: https://webrtc-codereview.appspot.com/387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:13:12 +00:00
stefan@webrtc.org
4b377414f1 Fix release build errors. 
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/394005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 13:50:57 +00:00
stefan@webrtc.org
9c84b0dc9f Fix build errors with GCC. 
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 13:14:04 +00:00
stefan@webrtc.org
7adab0922d This removes the knowledge of frame completeness from the FEC decoder. 
Therefore, with this change a recovered packet is only considered old,
and will be removed, if more than 48 recovered packets are stored.

Packets are immediately reconstructed and sent to the jitter
buffer. Before this CL packets were processed on a frame-by-frame
basis, and all packets belonging to a frame was sent to the
jitter buffer at the same time.

The number of FEC packets is also limited to 48. An FEC packet is
removed if all protected packets have been recovered or if the
FEC packet is considered old.

Lot's of tests added.

Patch-set 2:
- Fixed rtp_fec_unittest.cc to work with the new reconstruction.
- Added reference counting of Packet to be able to keep references to them from FecPacket between different reconstruction runs.
- Rewrote the packet search to use std::set_intersection.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 12:34:52 +00:00
henrike@webrtc.org
d5657c2f69 Refactored files according to google style since http://review.webrtc.org/314001/ is blocked on this and formatting changes should not be done with code changes. 
Review URL: https://webrtc-codereview.appspot.com/387005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 23:41:49 +00:00
andrew@webrtc.org
68da6adafe Remove WebRtc_ types. 
Allows us to avoid the "cast to UWord32" Coverity coverup.

BUG=
TEST=test_fec

Review URL: https://webrtc-codereview.appspot.com/379002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 22:24:14 +00:00
tina.legrand@webrtc.org
df69775bfa Adding support for full-stereo codec. 
This is an experiment, letting Celt represent a full-stereo codec.

Review URL: https://webrtc-codereview.appspot.com/379013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 10:22:21 +00:00
phoglund@webrtc.org
78088c2f36 Removed warnings on Windows and enabled warnings-as-errors on Windows. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:56:45 +00:00
mflodman@webrtc.org
657b2a4965 Added return due to gcc complaints in r1604. 
TBR=andrew

TEST=Bulid with clang version 3.1 (trunk 148911) and gcc.

Review URL: https://webrtc-codereview.appspot.com/384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 11:06:01 +00:00
mflodman@webrtc.org
c80d9d9361 Removed default cases causing clang errors, -Wcovered-switch-default. 
BUG=
TEST=Bulid with clang version 3.1 (trunk 148911)

Review URL: https://webrtc-codereview.appspot.com/379008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 10:11:25 +00:00
andrew@webrtc.org
4942832928 Fix "may be used uninitialized" warning. 
TBR=marpan@webrtc.org
BUG=
TEST=build on Linux/Release and rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/381004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-04 05:23:51 +00:00
marpan@webrtc.org
b783a55df3 Unit test for forward_error_correction. 
Review URL: https://webrtc-codereview.appspot.com/358006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-04 02:46:35 +00:00
marpan@webrtc.org
307c1ff20c Fix for issue #254: windows crash of test_fec. 
Review URL: https://webrtc-codereview.appspot.com/379010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-04 02:45:22 +00:00
stefan@webrtc.org
07b45a5dc4 Added API for getting the send-side estimated bandwidth. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/372006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-02 08:37:48 +00:00
andrew@webrtc.org
7fe219f681 Add some additional checks for corrupt payload. 
Investigation with corrupt payloads revealed a few places we could
be using stronger checks. These are not foolproof by any means, but
I figure the earlier we catch this the better.

BUG=242
TEST=loopback call with a hacked ViE to insert corrupt payloads, and vie_auto_test without the hacks.

Review URL: https://webrtc-codereview.appspot.com/369015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-01 02:40:37 +00:00
marpan@webrtc.org
cdba1a836b test_fec: Reduce execution time of test, and use testsupport/fileutils.h for path of randomSeedLog file. 
Review URL: https://webrtc-codereview.appspot.com/373016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 00:36:14 +00:00
pwestin@webrtc.org
5dad00be52 Coverty fix: FEC unintended signed extension and resource leaks. 
Review URL: https://webrtc-codereview.appspot.com/368010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1569 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 13:05:29 +00:00
stefan@webrtc.org
2fd1e1e40d Add unittests for ReceiverFec. 
Also added mock for RtpReceiverVideo and did appropriate changes to
allow for mocking.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 09:03:37 +00:00
pwestin@webrtc.org
04cf69a714 Coverty: cleanup CheckCSRC. 
Review URL: https://webrtc-codereview.appspot.com/369014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 13:47:19 +00:00
phoglund@webrtc.org
d1a860b415 Fixed GCC 4.6 errors (mostly 'unused variable' errors and incorrect usage of EXPECT_EQ with booleans. 
Fixed remaining compilation errors in release, etc.

Fixed errors from GCC 4.6 compilation.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/366008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-26 14:49:28 +00:00
henrike@webrtc.org
567b99be5f Coverity report: fixes an issue where the returnvalue of a function is not checked. 
Review URL: https://webrtc-codereview.appspot.com/347013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 23:43:54 +00:00
pwestin@webrtc.org
f6bb77a6f0 Cleaning up all use of RTP_PAYLOAD_NAME_SIZE and RTCP_CNAME_SIZE also fixed the char handing in trace. 
Review URL: https://webrtc-codereview.appspot.com/358001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 17:16:59 +00:00