hta@webrtc.org
fcb7c38b15
Two more sleep calls converted to use SleepMs().
...
This is CL 753005 in its new home.
BUG=603
Review URL: https://webrtc-codereview.appspot.com/1201008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3761 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-04 08:56:34 +00:00
henrika@webrtc.org
bb8ada686e
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
...
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled
Review URL: https://webrtc-codereview.appspot.com/1201010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-04 08:39:09 +00:00
pwestin@webrtc.org
0c45957e3a
Remove UDP transport API from VoE
...
Review URL: https://webrtc-codereview.appspot.com/1236004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-03 15:43:57 +00:00
henrika@webrtc.org
0746ce1465
Fixes memory leak in AudioLevel class reported by memory try bots.
...
TBR=tommi
Review URL: https://webrtc-codereview.appspot.com/1275008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3756 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-03 11:58:12 +00:00
henrika@webrtc.org
d108a46206
Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer
...
BUG=225690
Review URL: https://webrtc-codereview.appspot.com/1269008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3755 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-03 11:25:31 +00:00
pwestin@webrtc.org
82dcc9ff11
Remove UDP transport API from ViE
...
Review URL: https://webrtc-codereview.appspot.com/1232004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 20:37:14 +00:00
pbos@webrtc.org
7b859cc1e9
Webrtc_Word32 => int32_t in video_coding/main/
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 15:54:38 +00:00
henrike@webrtc.org
cfc07c943f
Revert of r3747.
...
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1277005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 14:55:44 +00:00
hta@webrtc.org
95d88735ee
Two more sleep calls converted to use SleepMs().
...
BUG=603
Review URL: https://webrtc-codereview.appspot.com/753005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3751 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 14:46:33 +00:00
henrika@webrtc.org
4ff956f428
Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer
...
BUG=159112
Review URL: https://webrtc-codereview.appspot.com/1201007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3750 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 11:59:11 +00:00
kjellander@webrtc.org
46e626d3b8
Fix gflags compile error on x86 Android
...
This CL is the landing of http://review.webrtc.org/1277004/ for yujie.mao@intel.com .
I verified the added files are identical with the previously added ones
in third_party/google-gflags/gen/arch/linux/ia32 (which is the way this library needs to be handled when supporting the additional Android platforms).
BUG=none
TEST=Successfully compiled WebRTC on Linux Precise with:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug
Review URL: https://webrtc-codereview.appspot.com/1273005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3749 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 11:07:04 +00:00
justinlin@chromium.org
f81fad6267
Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher
...
than 2^16kbps.
Review URL: https://webrtc-codereview.appspot.com/1275004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3748 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-01 22:25:11 +00:00
fbarchard@google.com
747c4cc96e
For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled.
...
BUG=none
TEST=run a hangout and screencast high framerate, high resolution windows of youtube. Observe that 1 cpu is insufficient to maintain high framerate with complex content.
Review URL: https://webrtc-codereview.appspot.com/1203006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3747 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-01 22:16:45 +00:00
elham@webrtc.org
65243bdb18
Updated Webrtc version to 3.28
...
Review URL: https://webrtc-codereview.appspot.com/1272006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3745 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-01 16:17:26 +00:00
marpan@webrtc.org
7f6b7cbcfc
Revert r3743.
...
TBR=andrew@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1272005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3744 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-29 21:35:22 +00:00
marpan@webrtc.org
e882a47c8d
Roll libvpx to 191157.
...
-Pick up the libvpx roll to 8015a9ae.
TBR=andrew@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1273004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3743 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-29 21:16:24 +00:00
marpan@webrtc.org
29f34b8727
Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549
...
Review URL: https://webrtc-codereview.appspot.com/1270004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3741 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-28 18:57:46 +00:00
henrike@webrtc.org
626c663115
Fixes build break in previous cl ( https://code.google.com/p/webrtc/source/detail?r=3739 ) found by Android bots.
...
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1269005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3740 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-28 16:31:51 +00:00
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