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Author SHA1 Message Date
pbos@webrtc.org
8911ce46a4 Generic video-codec support. 
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.

BUG=1442
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 16:39:03 +00:00
stefan@webrtc.org
41211466d8 Revert the deletion of test_api_nack.cc in r3674. 
TBR=mflodman@webrtc.org, mikhal@webrtc.org

BUG=1513

Review URL: https://webrtc-codereview.appspot.com/1217004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 15:00:50 +00:00
bjornv@webrtc.org
04ecd49ec5 Truncated delay quality to avoid negative return values 
This forces the output of last_delay_quality to the interval [0, 1] in Q14.

BUG=none
TESTED=audioproc_unittest, trybot

Review URL: https://webrtc-codereview.appspot.com/1211004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 14:15:12 +00:00
mikhal@webrtc.org
bda7f305c5 Adding RTX on source 
Review URL: https://webrtc-codereview.appspot.com/1190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 23:21:52 +00:00
tina.legrand@webrtc.org
73222cff1a Adding Opus frame length test 
BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 13:29:17 +00:00
kma@webrtc.org
33f22d01f0 Fixed a crash issue in NSX module. 
Run time error message for function WebRtcNsx_PrepareSpectrumNeon():  "Bad access at:  0x4f535c:  vst1.16{d16, d17, d18, d19}, [r2], r12"

Cause: "anaLen" was defined as int16_t and should have been read as such in assembly function WebRtcNsx_PrepareSpectrumNeon().

Fix: Changed anaLen's definition to int in the header file instead.

BUG=b/8382174
Review URL: https://webrtc-codereview.appspot.com/1202004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-14 21:44:12 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665 
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. 
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 17:52:42 +00:00
stefan@webrtc.org
2baf5f5fa0 Refactor webrtc specific Event implementation to an EventFactory. 
Review URL: https://webrtc-codereview.appspot.com/1187005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 08:46:25 +00:00
turaj@webrtc.org
b7edd06530 Remove DTMF detection. Talk team has been in the loop and there is no need for 
DTMF detection at the receiver side.

test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 22:27:27 +00:00
kma@webrtc.org
d6cd64ac6a Change intrinsic code in isac fix to let it pass chrome clang compiler. 
Compiler complains about variables not initialized in instructions veor_s32() and vset_lane_s32().
Review URL: https://webrtc-codereview.appspot.com/1187006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 17:45:41 +00:00
stefan@webrtc.org
03e3117d87 Removed redundant VP8 width/height and made sure the generic width/height is set. 
Review URL: https://webrtc-codereview.appspot.com/1158005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 09:59:27 +00:00
dwkang@webrtc.org
7473f89f63 Revert "Internal clean up: removing unused include line." 
(reverting https://webrtc-codereview.appspot.com/1177004)

BUG=none

Review URL: https://webrtc-codereview.appspot.com/1181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 01:43:00 +00:00
dwkang@webrtc.org
25316b2a09 Internal clean up: removing unused include line. 
BUG=none
TESTED=passed try server

Review URL: https://webrtc-codereview.appspot.com/1177004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 01:10:02 +00:00
kma@webrtc.org
e5a81ed793 Fixed issue 1497 in iSAC fixed point. 
Bit exact.
Review URL: https://webrtc-codereview.appspot.com/1177005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 00:23:21 +00:00
kma@webrtc.org
23da8622c0 Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform. 
Cycles of the whole iSAC codec was reduced by 7.9%, measured by offline file test, with time() function.

Bit exact.

** Code style cleanup is not considered in this CL. **
Review URL: https://webrtc-codereview.appspot.com/1069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-09 00:38:14 +00:00
kjellander@webrtc.org
971278a962 Splitting out video_coding_test executable again. 
This CL undoes the merge of the developer test tool and the gtest tests
that was merged in https://code.google.com/p/webrtc/source/detail?r=3176

Doing that, we get a pure gtest executable of
video_coding_integrationtests which can run properly on the bots.

BUG=none
TEST=Trybots passing + local execution on Linux with:
out/Debug/video_coding_integrationtests --gtest_print_time (to ensure it will be possible to run with runtest.py)

Review URL: https://webrtc-codereview.appspot.com/1171007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 10:20:53 +00:00
kma@webrtc.org
2951a6df4a Fixed an assembly code error in AECM for ARMv7. 
Possibly related to an AECM quality issue encountered at Chrome testing.
No bug was logged.
Review URL: https://webrtc-codereview.appspot.com/1160006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 18:25:34 +00:00
stefan@webrtc.org
84cd8e39cf Disable frame dropper for screenshare mode. 
BUG=1466

Review URL: https://webrtc-codereview.appspot.com/1170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 13:12:32 +00:00
stefan@webrtc.org
7c16c3c4a1 Move video_coding OWNERS to video_coding/. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 13:11:32 +00:00
andrew@webrtc.org
52b57cc0d5 Fix debug file buffer bug introduced in r3574. 
This correctly uses int16_t rather than float. Only affects the debug
file buffer, not the production code path.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1162008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 00:45:50 +00:00
bjornv@webrtc.org
91d11b3cdd Adds new AEC API to audio_processing. 
One unit test added.
Tested with audioproc_unittest and trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 16:53:09 +00:00
stefan@webrtc.org
1dc0aa2de2 Fix for build error on android introduced with r3609. 
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1164004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 09:30:47 +00:00
stefan@webrtc.org
a27107004d Split the NACK list into multiple RTCPs if it's too big. 
TEST=rtp_rtcp_unittests
BUG=1434

Review URL: https://webrtc-codereview.appspot.com/1148006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 09:02:06 +00:00
andrew@webrtc.org
f0a90c37c4 Expose the capture-side AudioProcessing object and allow it to be injected. 
* Clean up the configuration code, including removing most of the weird defines.
* Add a unit test.

Review URL: https://webrtc-codereview.appspot.com/1152005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 01:12:49 +00:00
bjornv@webrtc.org
7f95732fe2 AEC Refactoring: Removes lint warning 
Changed inlude order.

TBR=andrew@webrtc.org
TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1156004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 23:47:39 +00:00
stefan@webrtc.org
a64300af50 Refactor NACK list creation to build the NACK list as packets arrive. 
Also fixes a timer bug related to NACKing in the RTP module which could cause packets to only be NACKed twice if there's frequent packet losses.

Note that I decided to remove any selective NACKing for now as I don't think the gain of doing it is big enough compared to the added complexity. The same reasoning for empty packets. None of them will be retransmitted by a smart sender since the sender would know that they aren't needed.

BUG=1420
TEST=video_coding_unittests, vie_auto_test, video_coding_integrationtests, trybots

Review URL: https://webrtc-codereview.appspot.com/1115006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 15:24:40 +00:00
phoglund@webrtc.org
44f85a49d8 Fixed coverity defects (CID 14657 and 14656). 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1153006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 14:59:31 +00:00
fischman@webrtc.org
73ec386d8a VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. 
This saves ~15% CPU on a Nexus 7 running AppRTCDemo.

BUG=1169

Review URL: https://webrtc-codereview.appspot.com/1150005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-03 17:28:03 +00:00
andrew@webrtc.org
6be1e934ad Properly error check calls to AudioProcessing. 
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.

BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 18:47:28 +00:00
bemasc@google.com
603ae3ece2 Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. 
This CL changes the return code of these methods to indicate
success instead of failure when there is nothing to change.

This change appears to resolve an issue where enabling the
timestamp offset extension via SDP would result in a failure if
that extension had already been enabled.
Review URL: https://webrtc-codereview.appspot.com/1118008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 17:03:02 +00:00
andrew@webrtc.org
78693fe37c Return an error when greater than 16 kHz is used with AECM. 
BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1146005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 16:36:19 +00:00
marpan@webrtc.org
7d052c3cb2 Turn off error concealment in videoprocessor_integration tests. 
Review URL: https://webrtc-codereview.appspot.com/1123006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 16:44:28 +00:00
braveyao@webrtc.org
6b6eb44cca Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. 
ISSUE=529
TEST=unittest
Review URL: https://webrtc-codereview.appspot.com/1120006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 10:08:02 +00:00
stefan@webrtc.org
9e254133ad Rewrite the jitter buffer statistics test and put make it robust under valgrind. 
BUG=1158

Review URL: https://webrtc-codereview.appspot.com/1116008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 08:45:23 +00:00
bjornv@webrtc.org
132c15de30 AEC Refactoring: 
* Adds pointer to low level AecCore struct.
* Adds a simple unit test of this new call.

Tested with audioproc_unittest, trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1121006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 21:03:41 +00:00
stefan@webrtc.org
e1c4ed958d Fix to send a full NACK list at least roughly once every 1.5 x RTT. 
BUG=1434

Review URL: https://webrtc-codereview.appspot.com/1111007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 16:23:06 +00:00
kma@webrtc.org
83561fb173 Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. 
No written bug report.

Tested with Nexus-S. Issue disappeared with the change.
Review URL: https://webrtc-codereview.appspot.com/1126006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 01:16:44 +00:00
andrew@webrtc.org
91f325586d Refactor WebRtc_CreateBuffer to return the instance. 
Review URL: https://webrtc-codereview.appspot.com/1140005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 00:35:06 +00:00
andrew@webrtc.org
9fbd9ca849 Force a memcpy directly from the AEC ring buffer. 
Review URL: https://webrtc-codereview.appspot.com/1140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 23:31:56 +00:00
andrew@webrtc.org
ac1f877a5e Remove unneeded libvpx path from vp8 include_dirs. 
This is already provided by libvpx.gyp.

BUG=webrtc:1428

Review URL: https://webrtc-codereview.appspot.com/1139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 17:10:38 +00:00
andrew@webrtc.org
9ae1354e25 Refactor ring_buffer interface, add a feature and a test. 
* Add a RingBuffer typedef.
* Add the ability to force a memcpy by passing a null ptr. In some cases,
  we know we want a memcpy. This allows us to skip a potential
  intermediate memcpy.
* Add a stress test.

Review URL: https://webrtc-codereview.appspot.com/1111004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 17:07:35 +00:00
phoglund@webrtc.org
8a0662306d New attempt at fixing hard-coded libvpx source. 
TBR=ajm@webrtc.org
BUG=1428

Review URL: https://webrtc-codereview.appspot.com/1138004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 15:05:01 +00:00
phoglund@webrtc.org
9a6623b629 Revert "Fixing hard-coded libvpx source path." 
This reverts commit 1c603646da11971f13d66a75da31ebbb6eff37d9.

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1137004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 14:27:14 +00:00
phoglund@webrtc.org
8571c90c7f Fixing hard-coded libvpx source path. 
BUG=1428
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1127006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-25 14:09:12 +00:00
kma@webrtc.org
2f9bd247ad Ported assembly coding in APM from Android to iOS. 
Bugs=none
Test=trybots, and offline file bit-exact tests.
Review URL: https://webrtc-codereview.appspot.com/1066009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-23 04:16:59 +00:00
vikasmarwaha@webrtc.org
10987a851d Minor bug fix in maxFPS parameter declaration. 
TBR = perkj@webrtc.org,mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1123005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3559 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 22:03:39 +00:00
vikasmarwaha@webrtc.org
bf3a9b3cce Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero. 
Review URL: https://webrtc-codereview.appspot.com/1101013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 20:25:54 +00:00
bjornv@webrtc.org
60f83131e4 AEC refactoring: Moved typedefs to _internal.h 
* This was actually part of r3553
* Tested with audioproc_unittest, trybots

TBR=andrew@webrtc.org
TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1118005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 16:12:24 +00:00
tina.legrand@webrtc.org
7a7a008031 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

Committed: https://code.google.com/p/webrtc/source/detail?r=3543

Review URL: https://webrtc-codereview.appspot.com/1103012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-21 10:27:48 +00:00