The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.
Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.
BUG=webrtc:1395
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1355004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
Propose of this CL: Close the camera properly on MacOS in order to allow other apps to use it.
Changes in this CL:
1. video_capture_qtkit_info_objc.mm _captureDevicesInfo is never released. I have found this memory leak using Instruments from XCode. The patch is releasing it in dealloc.
2. In video_capture_qtkit_objc.h:
a) _captureDeviceName is not needed. Is allocated in the class but never used.
b) I don't see the role of the NSAutoreleasePool. also if you use it you have to release it when the class is destroyed. Otherwise you will leak memory. Libjingle has for each thread a pool on mac os.
3. In video_capture_qtkit_objc.mm
a) the camera is not stopped properly . See the changes from dealloc. NOTE : If you don't call [[_captureVideoDeviceInput device] close] other apps will not be able to use the camera since you are not closing your app
b) Removed QTCaptureDevice* videoDevice = (QTCaptureDevice*)[_captureDevices objectAtIndex:0]; I don't know why this because the desired camera is opened in setCaptureDeviceById and can be different than position 0 in the camera array. At this moment if you have two cameras and user want to pick the one on index 1 the app also locks the one on 0 .
Other changes I have done to improve (and are not in this CL):
a) I have set the FPS properly to the desired. I have succeeded to reduce the CPU with 3 % doing this. The current code for setting FPS is commented in webrtc
b) I have removed _rLock from the equation. I don't know if it's good or not but I hadn't understood what exactly we are trying to protect with this. Anyway in the current implementation is never released.
Review URL: https://webrtc-codereview.appspot.com/1097014
Patch from Silviu Caragea <silviu.cpp@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3886 4adac7df-926f-26a2-2b94-8c16560cd09d
When adding this in
we started getting linking problems on the mac_asan bot due to
the empty list of source files for the library target.
Please re-add it into modules.gyp when the library has source files
to compile.
BUG=none
TEST=Passing mac_asan trybot.
Review URL: https://webrtc-codereview.appspot.com/1313009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3882 4adac7df-926f-26a2-2b94-8c16560cd09d
The script previously only executed build/gyp_chromium to
generate projects. This is not sufficient since the
supplement.gypi has not yet been created (happens in
gclient runhooks). Therefore this CL changes it to run
gclient runhooks instead (marginally slower).
Otherwise the bots would have to generate the projects
twice, which is inefficient.
Annotations are also added to make the upcoming buildbot
steps more readable when the script is running as an
annotated step.
TEST=local compilation on OS X.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1342005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3878 4adac7df-926f-26a2-2b94-8c16560cd09d
It turns out the default shell becomes /bin/sh on Lucid. By specifying the shell for subprocess.check_call we ensure bash is used.
Thanks to yujie.mao@intel.com for pointing this out.
BUG=1659
TEST=Successful build with build_demo.py both on Ubuntu Lucid and Precise.
TBR=leozwang
Review URL: https://webrtc-codereview.appspot.com/1343004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3875 4adac7df-926f-26a2-2b94-8c16560cd09d
TBR=perkj
BUG=227036 (in crbug.com)
TEST=out\Debug\voe_auto_test.exe --automated --gtest_filter=Dtmf* where I
manually modified the test and used 100 as new PT (which I first verified was
already used by CN, 48000).
BUG=
Review URL: https://webrtc-codereview.appspot.com/1319010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3859 4adac7df-926f-26a2-2b94-8c16560cd09d
- Tests video_rtp_play.cc, video_rtp_play_mt.cc, decode_from_storage.cc rewritten
- rtp_player.cc/.h rewritten; added interfaces for externally setting up sinks
- Support for reading .rtp files pulled out into rtp_file_reader namespace
- Added support for reading .pcap (libpcap/wireshark/tcpdump) files, see pcap_file_reader
BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1201009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3856 4adac7df-926f-26a2-2b94-8c16560cd09d
