admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
3,311 Commits 1 Branch 0 Tags
Commit Graph

308 Commits

Author SHA1 Message Date
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
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