Commit Graph

6455 Commits

Author SHA1 Message Date
95a8ddd272 Android: rename android_build_type gyp variable.
Following Chromium r187556 this variable has been renamed to
android_webview_build to better describe what it does.

Contributed by torne@chromium.org (https://webrtc-codereview.appspot.com/1195006/).
Review URL: https://webrtc-codereview.appspot.com/1214005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3686 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-18 21:41:04 +00:00
f1ea0df728 Updated WebRTC version number to 3.26
Review URL: https://webrtc-codereview.appspot.com/1219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3683 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-18 17:45:04 +00:00
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
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
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
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
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
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
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
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
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
2dc0367406 Added destructors for tests to control destruct order
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1197005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3667 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 21:36:10 +00:00
15960c2b67 Increasing size of nack list in buffered mode.
Review URL: https://webrtc-codereview.appspot.com/1187007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3666 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 20:52:49 +00:00
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
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
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
728b7ea245 Tool found: pass by value when pass by reference is better in system wrapper unit test.
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1186006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3662 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-12 21:49:56 +00:00
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
23875c1694 Fixes issue detected by tool.
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3659 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-12 17:36:54 +00:00
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
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
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
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
3de314615f Fix frame_editing_unittest reference file handling.
This test was initializing strings for reference video files in a static
context, which makes is against the style guide and also makes the paths
become invalid when the test is launched from a working directory
outside the checkout.

Moving the initialization into the test fixture solves this.

BUG=none
TEST=Local execution launched from a directory outside the checkout tree on Win, Mac and Linux + trybots (for
compilation as they don't yet run the tools_unittests).

Review URL: https://webrtc-codereview.appspot.com/1178004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3647 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-11 15:30:33 +00:00
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
927296fd1b Lazy capture_device_info acquisition.
BUG=1484

Review URL: https://webrtc-codereview.appspot.com/1169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3641 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-08 13:12:29 +00:00
38ebf98c2a Refactor barcode decoder to use Zxing's C++ version
By using the C++ version of Zxing, we can avoid having Java and Ant
as a dependency when running Video quality analysis on the bots.
This makes it far more easy to setup automation on new machines.

I also moved the scripts into the webrtc/ folder so it will be synced by default when building in Chrome (eliminating the need of a separate solution).

This CL also removes the need of the FFMPEG_HOME variable and replaces
its use with a command line flag to make the tool run smoothly on
Windows.

BUG=none
TEST=locally running the script on Windows, Mac and Linux.

Review URL: https://webrtc-codereview.appspot.com/1099007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3640 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-08 10:58:21 +00:00
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
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
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
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
5d37139374 Fixed a ton of Python lint errors, enabled python lint checking.
BUG=

Review URL: https://webrtc-codereview.appspot.com/1166004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3627 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-07 09:59:43 +00:00
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
efe4edb6da Enabling bufffering mode with no sync module or VoE
BUG= 1454

Review URL: https://webrtc-codereview.appspot.com/1149006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3625 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-06 23:29:33 +00:00
a9a1df0035 Remove the error return on SetAGC failure introduced by r3605.
BUG=webrtc:1464

Review URL: https://webrtc-codereview.appspot.com/1166005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3616 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-05 23:36:10 +00:00
90eb5c84f9 1. Updated test pages to include Chrome Frame meta tag
2. Updated test pages to use adapter.js
Review URL: https://webrtc-codereview.appspot.com/1142004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3614 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-05 19:53:01 +00:00
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
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
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
24045c5a02 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise.
bug=issue1370
test=trybots
Review URL: https://webrtc-codereview.appspot.com/1121007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-05 03:14:22 +00:00
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
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
ec6226eedc Updated version number to 3.25
Review URL: https://webrtc-codereview.appspot.com/1149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3600 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-04 18:05:56 +00:00
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
17b867ae00 compile fix for get_nprocs() with uClibc
BUG=

Review URL: https://webrtc-codereview.appspot.com/1150006
Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3598 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-04 15:09:03 +00:00
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
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
2412085bc1 Don't upsample the capture signal early.
* Remove the unneeded _mixingFrequency.
* Rename CheckForSendCodecChanges to better elucidate its function.
* Remove an unnecessary memcpy.

Upsampling should be done late in the chain. This is practically relevant
on mobile, where the capture rate is fixed at 16 kHz. When using Opus, the
signal was upsampled to 32 kHz and was no longer compatible with AECM, which only supports up to 16 kHz.

NEEDS_QA=true
TEST=run calls with a variety of capture device rates and codecs
BUG=chromium:178040,webrtc:1446

Review URL: https://webrtc-codereview.appspot.com/1146004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3594 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-02 00:14:46 +00:00
ea386147f1 Update integration tests for idempotent RTP header settings.
Review URL: https://webrtc-codereview.appspot.com/1152004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3593 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-01 23:43:14 +00:00
7d6f11302e Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools.
Review URL: https://webrtc-codereview.appspot.com/1153004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3592 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-01 23:01:14 +00:00