similar to: ogg player on a clie

If you have any of the devices listed below, please test with a recent -stable or -current. They will stop working in 4.9 and 5.2 although old behavior can _temporarily_ be enabled by adding "options DA_OLD_QUIRKS" to your kernel config. If I don't hear from anyone, they'll be going away permanently after the releases. -Nate #ifdef DA_OLD_QUIRKS /* Below a list of

We are wholesale supplier of DJ equipment here in Taiwan.We have various model in stock such as Guitar,Amps/Effects,Bass,Drums,Keyboards,Live Sound in stock.interested buyers should contact us at the below address yangmiaofenggear at yahoo.com yangmiaofenggear at hotmail.com Pioneer DJM 909 ...................$500USD PIoneer DJM 700 ...................$450USD Pioneer DJM 800

Hello, We are currently seeking to employ individual?s world wide. How would you like to make money by simply driving your car advertising for YAMAHA. How it works? Here?s the basic premise of the "paid to drive" concept: YAMAHA seeks people -- regular citizens,professional drivers to go about their normal routine as they usually do, only with a big advert for new "YAMAHA YZF-R6

A thought has been bugging me for a month or so, but I don't know where to look for the answer. Would it be possible to port the OpenSSH ssh client to PalmOS and other handheld OS'es like Symbian (EPOC) or PocketPC? Is the processor on PDA's fast enough? Would many libraries have to be ported aswell? I know that PalmOS and EPOC has an IP stack implemented.. Since the development

Background Info: OS ver: Ubuntu 8.04 desktop (Hardy) WINE ver: 0.9.59 Sound device #1: Onboard SiS (661FX chipset) Realtek AC'97 Sound device #2: Yamaha DS-1 PCI Problem Description: No matter what combination of settings I use in System > Preferences > Sound and Applications > WINE > Configure WINE > Audio Tab I seem to use, I cannot seem to force WINE to use only the

Hello all, I am currently writing an application for PalmOS handheld devices that organizes media files on a memory card and allows the user to easily create playlists. The tracks I am using for testing (totalling about 1.5 GB) are nearly all in Ogg Vorbis format. Throughout my testing I have had very little trouble in parsing the comment fields from mp3, asf and ogg files. That is, all

Can Wine be set up to allow "drag and drop" between Ubuntu 9.04 Nautilus 2.26.2 file manager and a Windows app running in Wine? The Windows application is the "Yamaha File Utility" v1.1.0 which allows MIDI files to be transferred to a Yamaha CLP-150 Clavinova digital piano. Unfortunately, the software relies on drag and drop for the file transfer. In native Windows, I would

Hi ! I'd like to connect phpgroupware to asterisk: when a user click on a phone number, his phone rings and he gets connected to the number he just clicked. I've tried by putting various files in /var/spool/asterisk/outgoing, without results (we are using SIP phones + CAPI channels). Is there a way to do that ? (If it's impossible (something impossible in *, LOL ?!?) I will create

configure.ac | 2 - doc/Makefile.am | 2 - libswfdec/swfdec_as_frame.c | 8 ++-- libswfdec/swfdec_audio.c | 1 libswfdec/swfdec_audio_event.c | 24 +++++-------- libswfdec/swfdec_audio_event.h | 3 + libswfdec/swfdec_shape_parser.c | 8 ++-- libswfdec/swfdec_sound.c | 1 test/sound/adpcm-2-2.swf.1.0.raw |binary

libswfdec-gtk/swfdec_source.c | 2 libswfdec/swfdec_marshal.list | 2 libswfdec/swfdec_player.c | 55 ++++++++++++---------- libswfdec/swfdec_player.h | 4 - libswfdec/swfdec_player_internal.h | 2 libswfdec/swfdec_sprite_movie.c | 70 +++++++++++++++-------------- libswfdec/swfdec_swf_instance.c

configure.ac | 2 - doc/Makefile.am | 1 doc/swfdec-sections.txt | 2 + libswfdec/swfdec_buffer.c | 12 +++++++ libswfdec/swfdec_movie_as_drawing.c | 49 ++++++++++++++++++++++++++++---- test/image/Makefile.am | 12 +++++++ test/image/drawing-zorder-5.swf |binary

Hi ! We have Cisco 7912 phones, and the doc says that I can "create up to four speed dial buttons on my phone using the Cisco CallManager". Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D

When I try to use the wineoss.drv sound driver, Wine hangs up my whole system (as described by me in Wine Bug #2579). From what I have learned, this is caused by a kernel bug, but I can't fix it as I only found Andreas Mohr's patch for SB16-compatible cards (http://lkml.org/lkml/2002/6/16/45), while I'm using a Yamaha sound card (ymfpci.c kernel driver). When I disable the sound in the

hi, i need some small sip video endpoint for cloud videoconference (like bluejeans) i have this idea VIDEO OUT TV or projector with HDMI VIDEO IN cameras with h264 hw enconding - http://downloads.element14.com/raspberry-pi-camera/ <http://downloads.element14.com/raspberry-pi-camera/> - logitech C920 - Creative Live! Cam Connect HD - ??? ENDPOINT - raspberry - miniPC linux + asterisk

I want to record audio in ADPCM format. According to the "show codecs" output of Asterisk, it looks like it supports adpcm. But I do not know what to tell the "RECORD FILE" directive in my AGI script. The RECORD FILE command usually has this form: RECORD FILE <filename> <format> <timeout> [BEEP] It records fine in WAV or GSM if I enter "wav" or

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

Dear All, I'm using LLVM2.9 for profiling basic block frequency. I'm using following commands. rdpatel55 at ubuntu:~$ llvm-gcc -emit-llvm -O0 -c -o adpcm.bc adpcm.c rdpatel55 at ubuntu:~$ llvm-gcc -emit-llvm -O0 -c -o rawcaudio.bc rawcaudio.c rdpatel55 at ubuntu:~$ llvm-link -o main.bc rawcaudio.bc adpcm.bc rdpatel55 at ubuntu:~$ opt -q -f -insert-edge-profiling -o main.inst main.bc

Hi, I had the similar messages after an upgrade then downgrade. I was able to fix by removing all indexes files. Thomas > Le 28 nov. 2018 ? 22:02, JCA <1.41421 at gmail.com> a ?crit : > > This happening when my Thunderbird client is trying to establish a connection with a Dovecot server. Some background first: > > 1) I am running Thunderbird 6.20.1 from a Linux