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1309 Commits

Author SHA1 Message Date
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +00:00
turaj@webrtc.org
4b1cd5c5c0 G722-stereo has been missing when creating AudioDecoder. 
Review URL: https://webrtc-codereview.appspot.com/1266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:42:48 +00:00
turaj@webrtc.org
4d06db557a NetEq4 fails if the first packets inserted in are out-of-band DTMFs. 
I had to take few steps to solve this issue. I have comments on places I made cahanges to clarify why I did the change.

   
Review URL: https://webrtc-codereview.appspot.com/1195004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 18:31:42 +00:00
stefan@webrtc.org
e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. 
TBR=pwestin@webrtc.org

BUG=1524

Review URL: https://webrtc-codereview.appspot.com/1261005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 17:01:48 +00:00
fischman@webrtc.org
add50b94a5 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. 
(required bumping minSdkVersion to 14)

This fixes a RuntimeException thrown on GalaxyNexus (but not N7, N4, or NS)
during startPreview() after the sequence of Start(), Stop(), Start(); seemingly
GN's OMX stack can't deal with parallel startPreview() & setPreviewDisplay() in
this situation.

Also:
- Only set the surface in the camera when valid
- Remove duplicate assignment
- Fix error check on voiceChannel allocation to account for multiple channel creation due to orientation change causing onDestroy()/onCreate() on the app, and rampant use of process-static holders for VoE data.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:48:34 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
solenberg@webrtc.org
d8a6e72057 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1232005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:02:30 +00:00
fischman@webrtc.org
0e3077ab1f Restart Android capture after orientation change. 
Also prevent an NPE on exit.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 22:08:51 +00:00
andrew@webrtc.org
c83a00ad49 Add some VoE and AudioProcessing mocks. 
Includes a bit of shared helpers in fake_common.h.

Review URL: https://webrtc-codereview.appspot.com/1221004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 21:20:38 +00:00
pwestin@webrtc.org
db4185664c Introduced pause and resume to the pacer 
Review URL: https://webrtc-codereview.appspot.com/1217007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 23:39:29 +00:00
pbos@webrtc.org
ae4e2b352b WebRtc_Word -> stdint in audio_coding/g711/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1223004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 13:38:29 +00:00
stefan@webrtc.org
836af79f58 Remove incorrect asserts. 
BUG=1527

Review URL: https://webrtc-codereview.appspot.com/1214006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 12:15:44 +00:00
pbos@webrtc.org
01b507a406 WebRtc_Word -> stdint in audio_coding/cng/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1222004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 11:28:42 +00:00
wu@webrtc.org
af33b62a72 Fix -Wstring-conversion warnings. 
Review URL: https://webrtc-codereview.appspot.com/1215006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-20 21:22:48 +00:00
vikasmarwaha@webrtc.org
455370d5b1 Thread safety issue fix in incoming_video_stream.cc. See issue 1465. 
Review URL: https://webrtc-codereview.appspot.com/1216009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-20 16:57:09 +00:00
pbos@webrtc.org
8685090060 Account for header inside I420Encoder::InitEncode. 
Also verify that the header is part of the received payload inside
I420Decoder::Decode.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1211005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 11:39:03 +00:00
stefan@webrtc.org
3d0b0d6902 Follow-up fix for r3681. 
TESTS=trybots and vie_auto_test
BUG=1514

Review URL: https://webrtc-codereview.appspot.com/1216006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 10:04:57 +00:00
kma@webrtc.org
31829a7baf Fixed initialization of SPL in echo_control_mobile. 
BUG=8403556 (a possible fix)
Review URL: https://webrtc-codereview.appspot.com/1220004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 00:25:01 +00:00
stefan@webrtc.org
f4944d49cf Fix framerate sent to account for actually sent frames. 
TESTS=trybots
BUG=1481

Review URL: https://webrtc-codereview.appspot.com/1195005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:04:52 +00:00
stefan@webrtc.org
abc9d5b6aa Change VCM interface to take target bitrate in bits per second. 
This also solves issue 1469.

TESTS=trybots
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1215004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:00:51 +00:00
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