generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
3,563 Commits 1 Branch 0 Tags 108 MiB
8c2e78b2de
Commit Graph

3563 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +00:00
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +00:00
wu@webrtc.org
80fccc29de Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." 
> Removed CPU APIs from VoEHardware. Code is now only used by test applications.
> 
> BUG=8404677
> 
> Review URL: https://webrtc-codereview.appspot.com/1238004

TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 23:38:21 +00:00
henrike@webrtc.org
4c138e8fca Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
BUG=8404677

Review URL: https://webrtc-codereview.appspot.com/1238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 21:23:42 +00:00
leozwang@webrtc.org
458194ba65 Fix broken audio. 
The problem was introduced in 3712, no need to external transport in
real test app, revert the change.

TBR=pwestin@webrtc.org
BUG=1539
Review URL: https://webrtc-codereview.appspot.com/1266005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:55:54 +00:00
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
henrike@webrtc.org
686001dd96 Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events). 
Note that this means that there is no new code. The code has been taken directly from condition_variable_win.cc/h compensating minimally to be able to split up the two code paths.

Tested by:
1) Disabling native implementation and send to try bots.
2) Only return native implementation (i.e. if native implementation returns NULL there will be a crash when using the condition variable) and send to try bots.
3) The final cl sent to trybots.
All tests pass.

The changes are due to static analyzer code complaints.

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:16:05 +00:00
andrew@webrtc.org
1b31c78e5f Remove VoE's default call in Trace::SetLevelFilter. 
This is an application level setting. Applying it here has the potential to override the application's preferences.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:09:48 +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
andrew@webrtc.org
0633cccb4f Alphabetize include order in fake_voe_external_media.h. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 01:57:24 +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
andrew@webrtc.org
b87cc85beb Refactor unittest trace printouts to a separate class. 
This allows other tests/tools which don't depend on TestSuite to reuse the functionality.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1245004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 16:23:37 +00:00
sjlee@webrtc.org
b4c441a785 Enable the below APIs for iOS. 
class VoEAudioProcessing
  int RegisterRxVadObserver();
  int DeRegisterRxVadObserver();
  int SetEcMetricsStatus();
  int GetEcMetricsStatus()
  int GetEchoMetrics();
  int GetEcDelayMetrics();

class VoENetEqStats
  int GetNetworkStatistics();

class VoEVolumeControl
  int SetChannelOutputVolumeScaling();
  int GetChannelOutputVolumeScaling();
Review URL: https://webrtc-codereview.appspot.com/1159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 11:12:20 +00:00
fbarchard@google.com
7b48cedc57 libyuv r618 roll. Includes new psnr tool for evaluating codec quality. 
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1241005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3718 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-24 02:58:00 +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
elham@webrtc.org
14c9909ef6 Updated WebRTC version to 3.27 
Review URL: https://webrtc-codereview.appspot.com/1235004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 21:59:19 +00:00
pwestin@webrtc.org
a078d5cc38 Bugfix for extended RTP/RTCP test 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 20:03:03 +00:00
pwestin@webrtc.org
26e35e1d06 Move the VIE tests to use external transport instead of the built in udp transport 
Review URL: https://webrtc-codereview.appspot.com/1216010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 19:21:27 +00:00
andrew@webrtc.org
c1ffd337f1 Add trace printouts to all unit tests. 
Unfortunately, this requires splitting system_wrappers_unittests out of system_wrappers.gyp to avoid a cyclic dependency.

TESTED=ran a few unit tests and observed printouts

Review URL: https://webrtc-codereview.appspot.com/1221006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:23 +00:00
marpan@webrtc.org
94bc4cf905 Add min and target bitrate to VideoCodec. 
Review URL: https://webrtc-codereview.appspot.com/1214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:08 +00:00
pwestin@webrtc.org
e30823911c Move the VoE tests to use external transport instead of the built in udp transport 
Review URL: https://webrtc-codereview.appspot.com/1223006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 16:12:57 +00:00
tina.legrand@webrtc.org
e86f43b02a Roll Opus 1.0.2 
BUG=issue1532

Review URL: https://webrtc-codereview.appspot.com/1229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 12:39:54 +00:00
hta@webrtc.org
3ed599adb5 Bandwidth stats display in constraints-and-stats. 
Also shows off the report type and ID field, and logs less useless info.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1212007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 08:48:16 +00:00
pwestin@webrtc.org
999e900fb6 Creating a copy of Udp transport under webrtc/test 
Adding a test namespace, updating the include paths and renamed folder name.
Review URL: https://webrtc-codereview.appspot.com/1203004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 16:38:05 +00:00
hta@webrtc.org
2cec0b1670 Cleanup nanosleep -> SleepMs 
Remove some leftover stuff

BUG=603
TEST=

Review URL: https://webrtc-codereview.appspot.com/672005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 14:02:29 +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
braveyao@webrtc.org
f354e1f587 Add audio/video only option in apprtc 
ISSUE = issue 1507
TEST  = 
Review URL: https://webrtc-codereview.appspot.com/1216007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-20 00:23:55 +00:00
vikasmarwaha@webrtc.org
ebf49da9b2 Url option to change the resolution. 
Review URL: https://webrtc-codereview.appspot.com/1218005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 22:15:55 +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
hta@webrtc.org
ecfd32880e Changed stats reporting to not use local/remote 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1216004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 08:45:47 +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
wjia@webrtc.org
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
elham@webrtc.org
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
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
kjellander@webrtc.org
3cb42b11bf Remove GCC 4.6 bot from LKGR parsing. 
As all Linux bots are gPrecise now, this bot is removed.

TEST=none
BUG=none
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1218004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 16:05:36 +00:00
pbos@webrtc.org
71335ce120 Have git ignore ".swp" files. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1210005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 15:21:51 +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