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Author SHA1 Message Date
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
hclam@chromium.org
4e16f25774 Remove atomicops.h from WebRTC 
atomicops.h are not necessary in trace_event.h similar to the port in WebKit.
It will cause a benign race condition detected by TSAN. If it shows up in
TSAN we will either suppress it or annotate it with dynamic annotations.

BUG=1215
Review URL: https://webrtc-codereview.appspot.com/982004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 01:13:19 +00:00
marpan@webrtc.org
9f0fc97d2f Rolllibvpx to 7a09f6b89268 
Relevant updates/fixes:

000c8414b510: Moved denoiser frame copy/updates out of loopfilter thread.
Multi-threading bug fix: http://code.google.com/p/webm/issues/detail?id=497

ef2248a2a376: Added work buffer for denoiser.
Denoiser bug fix: http://code.google.com/p/webm/issues/detail?id=485

464b1df6d45b: Updates to qp-regulate and rate correction factor.
Rate control improvement: http://code.google.com/p/webrtc/issues/detail?id=1153

TBR=andrew@webrtc.org, leozwang@google.com
Review URL: https://webrtc-codereview.appspot.com/981005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 23:02:32 +00:00
hclam@chromium.org
770a01e3b0 Fix build by including trace_event_internal in webrtc namespace 
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 22:51:38 +00:00
hclam@chromium.org
f222a00881 Use TRACE_EVENT to track time spent in VP8 encoding 
Using the TRACE_EVENT macro to log VP8 encoding events.
Review URL: https://webrtc-codereview.appspot.com/968011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 22:27:55 +00:00
kjellander@webrtc.org
d2bcde2e4e Suppressing TSan warnings for system_wrappers_unittests 
This CL makes system_wrappers_unittests pass on Lucid (it passed on Precise
without them for me).

BUG=300
TEST=Try job on linux_tsan
TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/976007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 21:39:18 +00:00
hclam@chromium.org
ad7efa6944 Port Chromium's trace_event.h to WebKit and add 
trace_event.h is ported from Chromium code.

These files are defined new for WebRTC:
* event_tracer.h
* event_tracer.cc
* event_tracer_unittest.cc
Review URL: https://webrtc-codereview.appspot.com/933034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 21:19:08 +00:00
kma@webrtc.org
02d9df4544 Updated webrtc_resources_revision to 11, for adding two test files for APM and iSAC. 
Review URL: https://webrtc-codereview.appspot.com/973014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 19:33:56 +00:00
stefan@webrtc.org
71258c594b Add a third full stack test and support for random jitter in ext transport. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/975005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 15:14:56 +00:00
mflodman@webrtc.org
eaf7cf26fe Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced. 
Review URL: https://webrtc-codereview.appspot.com/935032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 11:47:22 +00:00
kjellander@webrtc.org
f98ffc6db3 Removing default trybot names 
This is removing the default try bot names added in r3031. It doesn't seem like we need to avoid sending all jobs to these bots, even if they're much slower.

BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/978004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 10:00:41 +00:00
turaj@webrtc.org
42259e7ebc VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Committed: https://code.google.com/p/webrtc/source/detail?r=3231
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 02:15:12 +00:00
turaj@webrtc.org
36965b1803 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. 
TBR=tlegrand@google.com

TEST=voe_cmd_test, ACM unittest.
Review URL: https://webrtc-codereview.appspot.com/974011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 23:52:43 +00:00
marpan@webrtc.org
55edaecc93 Revert r3254 due to bot failure on android. 
TBR=andrew@webrtc.org, leozwang@google.com
Review URL: https://webrtc-codereview.appspot.com/971018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 19:59:03 +00:00
marpan@webrtc.org
1f3476dd83 Roll libvpx to 000c8414b510. 
Relevant updates/fixes:

000c8414b510: Moved denoiser frame copy/updates out of loopfilter thread.
Multi-threading bug fix: http://code.google.com/p/webm/issues/detail?id=497

ef2248a2a376: Added work buffer for denoiser.
Denoiser bug fix: http://code.google.com/p/webm/issues/detail?id=485

464b1df6d45b: Updates to qp-regulate and rate correction factor.
Rate control improvement: http://code.google.com/p/webrtc/issues/detail?id=1153
Review URL: https://webrtc-codereview.appspot.com/971017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 18:11:30 +00:00
phoglund@webrtc.org
5bbe069f28 Reformatted event* classes. 
TEST=Ran trybots.

Review URL: https://webrtc-codereview.appspot.com/972012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 10:44:37 +00:00