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Author SHA1 Message Date
brykt@google.com
f556890844 Added possibility to repeat frames. Also added unittest for that feature. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/994005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 11:42:45 +00:00
mflodman@webrtc.org
d73527ccab Changed assert to log. 
Review URL: https://webrtc-codereview.appspot.com/1010004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:26:17 +00:00
tina.legrand@webrtc.org
d0d41498a3 Adding AUDIO application as default for Opus stereo 
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.

I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.

Next step is to add an API to choose application mode.

BUG=issue1239

Review URL: https://webrtc-codereview.appspot.com/1007006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:23:10 +00:00
phoglund@webrtc.org
ad0ed582b5 Fixed a missed initialization (found by valgrind FYI bot). 
http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/327/steps/memory%20test%3A%20memcheck_voe_auto_test/logs/stdio

BUG=
TEST=Reproduced valgrind error, verified gone after fixing.

Review URL: https://webrtc-codereview.appspot.com/1008005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:14:36 +00:00
leozwang@webrtc.org
ac77084583 Roll opus to 172355 and delete opus_demo from webrtc opus 
opus_demo has been inlucded in opus in chromium.

BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/973013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 17:24:30 +00:00
phoglund@webrtc.org
6bc5d4dc07 Reformatted sort. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/998006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 14:55:24 +00:00
stefan@webrtc.org
1960219530 Make protection method, filename and resolution configurable for FullStackTest. 
Review URL: https://webrtc-codereview.appspot.com/991007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 12:45:16 +00:00
tina.legrand@webrtc.org
4275ab1ca0 Implement NetEq duration estimation for Opus. 
Review URL: https://webrtc-codereview.appspot.com/983004
Patch from Ralph Giles <giles@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 09:52:45 +00:00
leozwang@webrtc.org
515ef2428c Clean up variable after it gets deleted 
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/939038

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 05:36:36 +00:00
mikhal@webrtc.org
e239bf0940 Making I420VideoFrame ref-counted 
BUG=937
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 00:07:57 +00:00
kjellander@webrtc.org
b13dfbffd7 Making barcode tools work on Windows + fixes. 
This makes it possible to compile on the bots without hardcoding paths
to Ant, Java and ffmpeg deep into the Python scripts (hardcoded paths exists only in the buildbot configuration).
For bots, the ANT_HOME, JAVA_HOME and FFMPEG_HOME environment variables must be set to the install locations for each of these dependencies, for Windows.

This CL also improves the return code handling to make failures easier to detect when things break.

TEST=running build_zxing.py without Ant or Java in the PATH, but with
ANT_HOME, JAVA_HOME and FFMPEG_HOME set. Running Chromium's src/chrome/test/functional/webrtc_video_quality.py.
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1002005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 19:53:00 +00:00
mikhal@webrtc.org
0b18fb38e6 vie auto test: Adding a constructor for NetworkParameters 
BUG=

Review URL: https://webrtc-codereview.appspot.com/995013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 19:47:52 +00:00
mikhal@webrtc.org
622c8bd0cc ViE autotest: Adding loss models to the external transport 
Review URL: https://webrtc-codereview.appspot.com/1000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 17:21:51 +00:00
phoglund@webrtc.org
6e0ce73741 Reformatted map classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1006004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 17:18:35 +00:00
phoglund@webrtc.org
61f39a3174 Fixed bad header name. 
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1001008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 16:02:13 +00:00
phoglund@webrtc.org
07bf43c673 Replaced the _audio parameter with a strategy. 
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 15:40:53 +00:00
phoglund@webrtc.org
59ad541e57 Reformatted rw_lock classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1007004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 15:20:35 +00:00
stefan@webrtc.org
eaebeb36ae Without specifying the input files the offsets will not automatically be regenerated when building for different architectures. That is very risky as it will cause crashes rather than build errors. 
TEST=trybots

BUG=1185

Review URL: https://webrtc-codereview.appspot.com/975006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 08:38:50 +00:00
kjellander@webrtc.org
10abe25f6d Make audioproc output files be written to output dir by default. 
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat

TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1003005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-17 18:28:07 +00:00
fbarchard@google.com
3c37354b70 Initialize 3 variables which are preventing VS2012 from building. 
BUG=1211
TESTED=ninja -C out\Release
Review URL: https://webrtc-codereview.appspot.com/992005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-15 01:09:18 +00:00
fbarchard@google.com
4c32439830 Roll libyuv to r520. Includes security fix to mark stack as not executable. 
BUG=1172
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1000005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-15 00:20:08 +00:00
elham@webrtc.org
ad6845f4c4 Updated version number to 3.19 
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/995007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 21:28:09 +00:00
hclam@chromium.org
c5fcb0879b Update trace_event.h to match the one in Chromium 
Chromium's trace_event.h has updated to remove some not-well-used features.
Update WebRTC's copy to match.
Review URL: https://webrtc-codereview.appspot.com/995006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 21:16:46 +00:00
fbarchard@google.com
dec09eed2f libyuv r515 ports matrix effects to Neon 
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/966034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 15:22:25 +00:00
mflodman@webrtc.org
4aee6b637d Added API to get receive side video delay. 
BUG=1222

Review URL: https://webrtc-codereview.appspot.com/997004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 14:02:10 +00:00
phoglund@webrtc.org
1c75918302 Disabled flaky test. 
From flake in http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/270

Review URL: https://webrtc-codereview.appspot.com/1001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 10:40:05 +00:00
phoglund@webrtc.org
7659d914bb Decoupled video rtp receiver from rtp receiver. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/995005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:57:37 +00:00
phoglund@webrtc.org
52d981f60c Reformatted list classes. 
BUG=
TEST=Trybots

Review URL: https://webrtc-codereview.appspot.com/995004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:52:34 +00:00
stefan@webrtc.org
32519398b6 Remove latency excl network and add render time diff stats. 
Review URL: https://webrtc-codereview.appspot.com/996004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:03:27 +00:00
roosa@google.com
b8ba4d8109 Add number of inserted samples to NetEq statistics. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/964030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 00:06:18 +00:00
turaj@webrtc.org
c454fab03b Reformatting ACM. All changes are bit-exact in this CL. 
TEST=VoE auto-test, audio_coding_module_test; 

only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision
Review URL: https://webrtc-codereview.appspot.com/937035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 22:46:43 +00:00
elham@webrtc.org
ddebc17bee Fix for buffer overflow, WebRTC issue 1196 
Review URL: https://webrtc-codereview.appspot.com/998004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 21:55:47 +00:00
mikhal@webrtc.org
96dc6270d4 vpm unit test: Diasble frame dropping in tests 
(follow up on r3284)

BUG=

Review URL: https://webrtc-codereview.appspot.com/991005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 19:53:26 +00:00
mikhal@webrtc.org
4493db5a3e vpm: removing unnecessary memcpy 
TEST=trybots

BUG=1128

Review URL: https://webrtc-codereview.appspot.com/966038

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 18:25:36 +00:00
mflodman@webrtc.org
7acb65a870 Added jitter to fake network pipe. 
Review URL: https://webrtc-codereview.appspot.com/988004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 15:53:11 +00:00
stefan@webrtc.org
91c91df35a Track the actual render time rather than the decode time. 
Review URL: https://webrtc-codereview.appspot.com/993004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 15:26:01 +00:00
brykt@google.com
e19b078ebe Changed so that frame_cutter takes and argument where one can specify in which interval the frames should be deleted between the first frame to cut and the last frame to cut. This can for example be used to decrease the frame rate. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/966037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 14:46:40 +00:00
kjellander@webrtc.org
0240e8e90f Wider TSAN suppression for issue 300 
On some machines, this test has still been failing, so I'm widening the
suppression to resolve this.

BUG=300
TEST=passing linux_tsan trybot.

Review URL: https://webrtc-codereview.appspot.com/992004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 13:02:29 +00:00
phoglund@webrtc.org
92bb417cb1 Decoupled RTP audio processor from RTP receiver. 
BUG=
TEST=Ran vie_auto_test, rtp_rtcp_unittests, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 10:48:24 +00:00
phoglund@webrtc.org
5b689efe8e Will now only require near-perfect PSNR and SSIM. 
BUG=
TEST=Ran test and checked we accept somewhat lower values.

Committed: https://code.google.com/p/webrtc/source/detail?r=3269

Review URL: https://webrtc-codereview.appspot.com/964031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 10:15:06 +00:00
fbarchard@google.com
86464eacb6 ISAC_main_inst initialized to NULL to avoid potentially garbage pointer passed to WebRtcIsacfix_EncoderInit 
BUG=1211
TESTED=local build on Windows.  Failed previously with vs2012.  With this change kenny.cc builds.
Review URL: https://webrtc-codereview.appspot.com/984004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 07:47:54 +00:00
mikhal@webrtc.org
a8544eaf03 Vp8 tests: Removing legacy unused tests and reorganization of existing ones. 
Review URL: https://webrtc-codereview.appspot.com/972013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 00:37:22 +00:00
kma@webrtc.org
7877b0f6d2 Added noexecstack markers for assembly files (webrtc issue 1172). 
Webrtc builds on ios, linux, android and other major platforms passed. Didn't do chrome build test.
Review URL: https://webrtc-codereview.appspot.com/987004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:22:13 +00:00
kma@webrtc.org
fa5b6bf4f4 Optimized WebRtcIsacfix_Spec2Time() for iSAC-Fix in ARM Neon processor. Speed doubled. 
Review URL: https://webrtc-codereview.appspot.com/930033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:00:52 +00:00
roosa@google.com
1b60ceb499 Add GetAudioFrame API to VoiceEngine. 
Allows the caller to pull frames from a channel instead of sending them to the output mixer.

BUG=

Review URL: https://webrtc-codereview.appspot.com/973012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:00:29 +00:00
roosa@google.com
b718619f0a Expose NetEq playout mode off through VoiceEngine. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/971016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 21:59:14 +00:00
roosa@google.com
0870f02cdb Add API to retreive last received RTP timestamp to VoiceEngine. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/969016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 21:31:41 +00:00
andrew@webrtc.org
d8aeb30d55 Revert 3269 
> Will now only require near-perfect PSNR and SSIM.
> 
> BUG=
> TEST=Ran test and checked we accept somewhat lower values.
> 
> Review URL: https://webrtc-codereview.appspot.com/964031

TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 20:58:32 +00:00
phoglund@webrtc.org
735a6cec96 Will now only require near-perfect PSNR and SSIM. 
BUG=
TEST=Ran test and checked we accept somewhat lower values.

Review URL: https://webrtc-codereview.appspot.com/964031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 15:20:34 +00:00
phoglund@webrtc.org
740be44af5 Reformatted file_* classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/980004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 12:52:15 +00:00