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Author SHA1 Message Date
phoglund@webrtc.org
a22a9bd9ca Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. 
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-14 10:01:55 +00:00
andrew@webrtc.org
bafdae3cfc Fix simulated analog gain in audioproc. 
* It doesn't make much sense to apply at all when reading from the protobuf.
* Reduced the gain to be closer to actual mics.

BUG=1260

Review URL: https://webrtc-codereview.appspot.com/1027007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 23:11:29 +00:00
andrew@webrtc.org
f908011eb4 Remove extra line. 
TBR=elham

Review URL: https://webrtc-codereview.appspot.com/1024008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 22:39:55 +00:00
marpan@webrtc.org
ef1a760446 Rounding error fix in media_opt_util. 
Review URL: https://webrtc-codereview.appspot.com/1013006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 22:13:19 +00:00
mflodman@webrtc.org
2f225cadde Add logs when no RTCP RR has been received for three regular RTCP intervals. 
BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 13:54:43 +00:00
mikhal@webrtc.org
658d423e81 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers. 
BUG=988

Review URL: https://webrtc-codereview.appspot.com/995014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-08 19:19:59 +00:00
phoglund@webrtc.org
c38eef896a Reformatted RTPReceiver. 
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)

BUG=
TEST=Trybots, vie_ & voe_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/998007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-07 10:18:30 +00:00
stefan@webrtc.org
1ea4b502ef Refactor receiver.h/.cc. 
TEST=video_coding_unittests, vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/994008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-07 08:49:41 +00:00
kma@webrtc.org
f545cf8f10 Addressing webrtc issue 1237, http://code.google.com/p/webrtc/issues/detail?id=1237. 
Code compared to C. Bit-exact.
Review URL: https://webrtc-codereview.appspot.com/1021004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-04 17:40:21 +00:00
andrew@webrtc.org
00c7c4315b Replace voice engine utility functions with system wrapper variants. 
SLEEP -> SleepMs
GET_TIME_IN_MS -> TickTime::MillisecondTimestamp

These could cause unused function errors on some compilers.

BUG=1228

Review URL: https://webrtc-codereview.appspot.com/1013004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 16:06:39 +00:00
pwestin@webrtc.org
1b6da28047 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. 
Landing of 573005 On behalf of an1kumar@gmail.com

TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-21 17:46:24 +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
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
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
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
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
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
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
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
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
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
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
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
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
braveyao@webrtc.org
72feb0b2e2 Not to enum NOTPRESENT audio devices with CoreAudio on Win 
BUG = 
TEST = Manual test
Review URL: https://webrtc-codereview.appspot.com/939037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 22:36:07 +00:00
mikhal@webrtc.org
451aa5dd9d Adding vp8 sequence coder: simple command line encode and decode. 
Goal is to replace existing normal test and affiliates (will be done in follow up cl's)
BUG =1070

Review URL: https://webrtc-codereview.appspot.com/935029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 21:23:02 +00:00
andrew@webrtc.org
3a5a8a8bcc Properly zero out unmixed frames. 
BUG=6770157

Review URL: https://webrtc-codereview.appspot.com/933037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 19:37:16 +00:00
kma@webrtc.org
0e739508e0 Added buildbot benchmarking in iSAC and APM into Android platform build. 
Review URL: https://webrtc-codereview.appspot.com/964022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 15:26:28 +00:00
mikhal@webrtc.org
b968213f3c vp8 test: Updating creation of enc/dec 
Review URL: https://webrtc-codereview.appspot.com/937036

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 19:14:26 +00:00
mikhal@webrtc.org
251f64e9e8 Updating vp8 test structure 
Review URL: https://webrtc-codereview.appspot.com/935031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 17:56:20 +00:00
mikhal@webrtc.org
60d25f90ff Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. 
This is a follow up on r3227.

Review URL: https://webrtc-codereview.appspot.com/929038

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 16:48:18 +00:00
henrik.lundin@webrtc.org
75f8c78d08 Fixing path to ptypes.txt in NetEqRTPplay 
The default path to the file ptypes.txt needed by NetEqRTPplay
had gone old. Updating to new repo layout.

Also purging old payload types from the file itself.

Review URL: https://webrtc-codereview.appspot.com/966035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-06 15:22:00 +00:00
turaj@webrtc.org
226db898f7 Dual-stream implementation, not including VoE APIs. 
Review URL: https://webrtc-codereview.appspot.com/933015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 22:13:31 +00:00
turaj@webrtc.org
277ec8e3f5 Fix a bug when iSAC-48kHz was added. 
I discovered this by running extended VoE test on "Codecs."

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/973010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 21:16:23 +00:00
mikhal@webrtc.org
f18de86db1 Revert 3227 
> vp8 unittest: Adding qcif stride test
> 
> Review URL: https://webrtc-codereview.appspot.com/930030

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 20:08:57 +00:00
mikhal@webrtc.org
ab83bb39ad vp8 unittest: Adding qcif stride test 
Review URL: https://webrtc-codereview.appspot.com/930030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 19:12:29 +00:00
turaj@webrtc.org
b0dff12d2b 48 kHz extension to iSAC. 
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 17:43:52 +00:00
stefan@webrtc.org
8d0cd07d0c Add test to verify that padding only frames are passing through the RTP module. 
Review URL: https://webrtc-codereview.appspot.com/934023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 14:01:46 +00:00
tina.legrand@webrtc.org
5b4fe494e7 Changing default bitrate to 64000 bps for Opus. 
Default settings for Opus in WebRtc is stereo, but we had default rate to 32 kbps. This is too low for stereo, where we need to encode using 64 kbps to get the quality we like.

BUG=

Review URL: https://webrtc-codereview.appspot.com/974008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 12:08:53 +00:00
kjellander@webrtc.org
ad0f3baf90 Removing redundant codec unittest targets. 
The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests

Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).

The following test has been removed since it was empty:
* audio_conference_mixer_unittests

BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)

Review URL: https://webrtc-codereview.appspot.com/971008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 10:52:29 +00:00
stefan@webrtc.org
c94f8d4e8f Fix OOB read in padding tests. 
BUG=1177

Review URL: https://webrtc-codereview.appspot.com/973009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 08:57:54 +00:00
henrike@webrtc.org
fc4a7ee807 Fixes chromium build bots. 
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/971014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 16:17:44 +00:00