eb7ebf20ed
Revert 3543
...
> Changing non-const reference arguments to pointers, ACM
>
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
>
> BUG=issue1372
>
> Review URL: https://webrtc-codereview.appspot.com/1103012
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 15:57:31 +00:00
374aa49e1a
Changing non-const reference arguments to pointers, ACM
...
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.
BUG=issue1372
Review URL: https://webrtc-codereview.appspot.com/1103012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 15:22:23 +00:00
aea96d36e3
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
...
to avoid ODR violations with peerconnectioninterface.h in libjingle.
Conflict introduced in
https://webrtc-codereview.appspot.com/1060005/diff/14010/webrtc/modules/interface/module_common_types.h#newcode326
TEST=none
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1105011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3540 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-19 22:09:36 +00:00
0a480cbe4d
Added getter for far_time_buf in AEC. Only used in AEC debug dump.
...
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1110005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3539 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-19 21:41:27 +00:00
5fc829200c
This refactoring CL moves the nlp_mode member value from aecpc_t to aec_t, since it it never used at that level. Further, I removed two suppression variables by depending on nlp_mode directly.
...
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1095007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3538 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-19 21:06:52 +00:00
cea70f4055
* Name change
...
* Removed WebRtcAec_ function name prepending on private function.
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1096012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3537 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-19 21:03:10 +00:00
95b48c3551
Update to codec unit test:
...
enable frame dropper for rate control test.
Review URL: https://webrtc-codereview.appspot.com/1099014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3536 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-19 20:02:32 +00:00
0460c7294a
Remove the dependency on dxguid.lib.
...
It turns out we don't really need it and therefore can also get rid of the added lib directory.
Review URL: https://webrtc-codereview.appspot.com/1094015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3530 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-19 12:13:03 +00:00
d2c3bed1da
Move directx_sdk_path definition variable into the video_render_module gyp file.
...
The variable is now:
* Only set and used for Windows (not globally for all platforms)
* Only used in the standalone build (include_internal_video_render == 1)
This means that we can remove the variable from Chrome and that the standalone
win builders should start picking up the local directx folder and turn green
(*crossesfingers*).
Review URL: https://webrtc-codereview.appspot.com/1103014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3529 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-18 14:53:04 +00:00
eb91792cfd
Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.
...
Review URL: https://webrtc-codereview.appspot.com/1105007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-18 14:40:18 +00:00
e3d6ffede4
Increase threshold in codec unit test.
...
Review URL: https://webrtc-codereview.appspot.com/1096011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3526 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:31:17 +00:00
ef9f76a59d
Adding a receive side API for buffering mode.
...
At the same time, renaming the send side API.
Review URL: https://webrtc-codereview.appspot.com/1104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00
47fe5736c1
Bug fix for webrtc issue 1391. Typo in sin_length for socket address.
...
Review URL: https://webrtc-codereview.appspot.com/1108004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3524 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 18:42:12 +00:00
b4cd342eb9
This refactoring CL contains an API to get low level echo metrics stats.
...
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1107007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3523 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 18:40:34 +00:00
21a2fc902d
This Cl includes
...
* A getter for echo_state
* Style changes, such as changes to int where appropriate
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1093011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3522 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 17:01:03 +00:00
325f625137
Moved the actual calculations to aec_core to avoid passing up low level members.
...
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1103011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3521 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 15:21:02 +00:00
6f6acd9f80
Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter.
...
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1099011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3517 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-14 21:17:12 +00:00
7267ffde56
Moved debug file handling to aec_core from echo_cancellation.c. This removes dependency on low level member variables.
...
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1093010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3515 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-14 17:56:23 +00:00
3e10249f20
Added delay estimation test to audio processing unit tests.
...
The test verifies that we get proper delay metrics when inserting delayed versions of the same file to far-end and near-end.
Failure of the test has been verified through a missmatch between AEC delay buffer size and test buffer size.
Also added a missing file rewind to another test and removed some lint warnings.
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1100004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3514 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-14 15:29:09 +00:00
a092cbf9b7
Fixing lint warnings from previous commit
...
In this CL I have removed (almost) all lint warnings I got for this commit:
https://code.google.com/p/webrtc/source/detail?r=3454 .
The only warning not fixed is a warning about usage of non-const reference. This will be fixed in a separate CL.
BUG=issue1372
Review URL: https://webrtc-codereview.appspot.com/1091006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3510 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-14 09:28:10 +00:00
9c4e662ea8
Fix Windows x64 errors in video_codecs_test_framework
...
Fixed a few size_t converted to int warnings (interpreted as errors).
Fixed cpplint warnings.
BUG=webrtc:1323
TEST=manual compile on Windows with GYP_DEFINES=target_arch=x64 and
ninja -C out\Debug_x64 (also compiled with Release_x64)
Review URL: https://webrtc-codereview.appspot.com/1097011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3507 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-13 09:35:12 +00:00
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
57a0049e25
VCM: Removing frame drop enable from Reset call
...
BUG = 1387
Review URL: https://webrtc-codereview.appspot.com/1097010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3500 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 21:23:23 +00:00
00ab7cf4fd
Fix perf output for audioproc and iSAC fixed-point tests
...
The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1093007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 12:33:03 +00:00
0cb48a0a18
Set SingleStream BWE in unittests.
...
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1094004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 08:30:23 +00:00
63066f7200
Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage.
...
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1098010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3493 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 08:27:33 +00:00
3d305c64b4
Updates to send side streaming mode:
...
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.
Review URL: https://webrtc-codereview.appspot.com/1101007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-10 18:42:55 +00:00
b64732abfc
Fix Win64 build breakage
...
This is for landing https://webrtc-codereview.appspot.com/1096006/ by Justin Schuh.
Stable will be updated after this has landed.
Review URL: https://webrtc-codereview.appspot.com/1091008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3484 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-07 10:14:05 +00:00
d83b9fdf45
Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged.
...
Bugs=none
Test=trybots, and file bit-exact tests; passed.
Description of the bug: Neon registers q4-q7 not saved before calling the outside FFT routines in the assembly functions.
Review URL: https://webrtc-codereview.appspot.com/1097006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3480 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-06 23:53:13 +00:00
959da8d286
Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.).
...
Bugs: none
Review URL: https://webrtc-codereview.appspot.com/1072007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3473 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-05 20:46:55 +00:00
a7303bdfb5
Lint-cleaned video and audio receivers.
...
BUG=
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/1093004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3471 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-05 15:12:39 +00:00
244251a9cd
Moved almost all payload-related stuff to the payload registry.
...
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.
BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1078004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 13:23:07 +00:00
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
254d85af54
Exchange TRY by enumerating image formats in Linux video capture
...
ISSUE = issue 529
TEST = unittest on Linux
Review URL: https://webrtc-codereview.appspot.com/1066011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3463 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 07:53:53 +00:00
becf9c897c
Fix mismatch between different NACK list lengths and packet buffers.
...
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.
BUG=1289
Review URL: https://webrtc-codereview.appspot.com/1065007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 15:09:57 +00:00
b586507986
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
...
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.
BUG=1298
Review URL: https://webrtc-codereview.appspot.com/1060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 14:33:42 +00:00
46d90dcd74
Adding three frame sizes to Opus
...
Adding support for 10, 40 and 60 ms packet sizes for Opus.
BUG=issue1015
Review URL: https://webrtc-codereview.appspot.com/1086004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 14:20:06 +00:00
aaad6134b9
Implementing stereo support for G.722
...
This CL implements stereo support for G.722 through a new class
AudioDecoderG722Stereo derived from AudioDecoderG722.
Also implementing tests for G.722 stereo.
Review URL: https://webrtc-codereview.appspot.com/1073006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3452 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 11:49:28 +00:00
ac46c6dac3
Replaced relative path to reference from <(webrtc_root).
...
Changed to proper include paths in AECM and NSX.
Tested on trybots.
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1063014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3450 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 21:06:16 +00:00
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
c0ada864b2
fix for issue 281.
...
A reverse copy is removed. The index to src buffer could be -1, this happens very often. The reverse copy is not needed as the content of the destination is overwritten further down in "WebRtcIlbcfix_CbConstruct()"
Bug=issue281
TEST=manual test over 1600 files TIMIT database, all outputs are bit-exact with the ones generated from head revision. Local run of asan does not generate any warning.
Review URL: https://webrtc-codereview.appspot.com/1063013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3447 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 18:21:06 +00:00
119c67df36
Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value.
...
This cl also includes tests and some clean up.
Review URL: https://webrtc-codereview.appspot.com/1019007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3445 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 17:18:02 +00:00
e07c661a29
VP8: Making key frame interval a tunnable parameter
...
Review URL: https://webrtc-codereview.appspot.com/1070006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 16:37:13 +00:00
6e3968f62a
Fix NetEq4 unit tests for VS2012
...
This merges the changes from r3199.
Review URL: https://webrtc-codereview.appspot.com/1078010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3443 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 15:07:30 +00:00
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
ac59dba3f7
Adding iSAC-fb support
...
Adding tests, too.
Review URL: https://webrtc-codereview.appspot.com/1070011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3440 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 09:55:24 +00:00
73a702c979
This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware.
...
Review URL: https://webrtc-codereview.appspot.com/1061007
Patch from Gil Osher <gil.osher@vonage.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3437 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 21:18:31 +00:00
7ded92b71e
Re-committing r3428
...
TBR=ajm
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1066008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3436 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 16:16:59 +00:00
51f11eb5ae
Fixing problems in audio_decoder_unittests
...
The tests did not work in Release mode because of the asserts.
Review URL: https://webrtc-codereview.appspot.com/1062010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3435 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 13:00:33 +00:00
ddf981c789
Disable iSAC fix test in audio_decoder_unittests
...
The test AudioDecoderIsacFixTest.EncodeDecode was disabled since it
triggers a valgrind warning. The issue is tracked in
https://code.google.com/p/webrtc/issues/detail?id=1353
BUG=1353
Review URL: https://webrtc-codereview.appspot.com/1084004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3434 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 12:29:48 +00:00
