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
mikhal@webrtc.org
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
tina.legrand@webrtc.org
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