similar to: alsa - spdif

Wine and spotify used to work. Now it doesn't, and I don't know why. Main problem is that Wine doesn't give any information back regarding why audio fails, only that it fails. When using winecfg, opening the Audio tag, choosing Alsa Driver and pressing "Test sound", a pop up reports "Audio test failed!". Nothing more. Choosing OSS, switching to emulation, same

I've never had any issues with Wine and audio up until now. The machine is a HTPC which is connected to a TV via HDMI. Audio works perfectly in everything without changing any additional settings, but in Wine it just refuses to work. This machine has no other sound devices other than the HDMI audio output, the onboard audio is disabled. Initially I just got the ALSA message about no master

I have some problems with wave audio in Wine. It sais that my device does not support any format at all. I run the following Code: #include <windows.h> #include <cstdio> typedef int (*EnumDevicesCallback)(const WAVEOUTCAPS* caps,void* param_through); int enumDevices(EnumDevicesCallback onDevice,void* param_through) { unsigned int n_devs=waveOutGetNumDevs(); unsigned int i=0;

Hello Forum/List, usually sound works quite well for me in WINE, but now trying to get my BlueTooth headphone to work (as opposed to the laptop's builtin speakers) I've run into a problem. My guess is it has to do with the delay occurs when a sound begins to play over these headphones, even though watching a movie in MPlayer at the same time to prevent that delay hasn't helped either.

I recently switched to OSS from alsa (loving it so far!), but one of the biggest problems I had to work through before was getting my usb microphone to work with Ventrilo. Before, I used these registry options: [Software\\Wine\\Alsa Driver] "AutoScanCards"="N" "DeviceCount"="2" "DevicePCM1"="default"

hello, can I get out of wine more than 16 bit output? sound_file -> player -> wine_out, which is 24/32 to linux? somethink like that, I am audiophile :( thank you

Wine List, Just looking for ideas, here's my system configuration: Gentoo Linux 2.4.24 ALSA built as modules Creative Labs X-Fi 1.18 Driver with a few modifications per http://gentoo-wiki.com/HARDWARE_Creative_Labs_X-Fi Game I'm trying to get sound working on is WoW 2.4 winecfg shows no OSS or ALSA mixers, and none show up in the game either. Now I know, the first thing you're

Hi, I cannot get wine to recognize my audio device (Nvidia GT210 w/HDMI). No other sound card is enabled. The right device for sound is hw:0,3. Alsa's speakertest works fine. Pulseaudio is completely removed from my system. Also Wine's pulse package is removed even though the trace still has some references to pulseaudio. Winecfg displays only MIDI devices under ALSA. Direct Sound options

So I used to use the winepulse patch for wine. (Yes I like pulse). But since 1.3.25+ uses mmdevapi for audio, that patch no longer works. That author points to a wine multimedia git, however, after installing that I have no more sound in any wine application, and the winecfg test audio says audio failed. How do I get this new sound server working in wine? -- Sincerely, MacNean C. Tyrrell

The following is mostly for Nick, but perhaps there are others who might be interested in one part or another of the following... Nick, I think we're basically on the same page here. I'll first try to cover some of my basic ideas in general, then respond to your specific comments inline. For one thing, I have long believed that it is smart to keep analog audio circuits

I'm after the cheapest way to decode a flac stream with following criteria; - Transport from UPnP (DNLA) NAS using either WiFi or Cat5 networking - Toslink digital out for using existing HiFi DAC. - Low power consuption (eg just few Watts - it's just a decoder anyway). There seems a real lack of Flac players that are cheap and HiFi separates integratable like current-day CD players. All

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most

Hi I am trying to use opus for encoding AES (spdif) stream . I suppose that input samples should be 16 bits in opus but samples in AES stream can be at most 24 bits. Can opus accepts samples more than 16 bits without any modifications in source codes ? If answer is No then "How can I modify opus to accept these 24 bits samples ?" Best Regards Kazem Baadpie -------------- next

Moritz Grimm (gtgbr@gmx.net) wrote : > Michael Smith wrote: > > >For the sake of convenience, it's nice that Vorbis' VBR supports 48kHz > > >at decent quality already, but to get back to the start of the > > >discussion, not all people have the equipment to play it back. To bring > > >highest possible satisfaction to the audience, 44.1kHz still is

On Fri, 26 Aug 2016, Peter Griffin wrote: > This patch adds the DT node for the uniperif reader > IP block found on STiH407 family silicon. > > Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen at st.com> > Signed-off-by: Peter Griffin <peter.griffin at linaro.org> > --- > arch/arm/boot/dts/stih407-family.dtsi | 26 ++++++++++++++++++++++++++ > 1 file

On Fri, 26 Aug 2016, Peter Griffin wrote: > This patch adds the DT node for the uniperif reader > IP block found on STiH407 family silicon. > > Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen at st.com> > Signed-off-by: Peter Griffin <peter.griffin at linaro.org> > --- > arch/arm/boot/dts/stih407-family.dtsi | 26 ++++++++++++++++++++++++++ > 1 file

Hi I am trying to use opus for encoding AES (spdif) stream . I suppose that input samples should be 16 bits in opus but samples in AES stream can be at most 24 bits. Can opus accepts samples more than 16 bits without any modifications in source codes ? If answer is No then "How can I modify opus to accept these 24 bits samples ?" Best Regards Kazem Baadpie -------------- next

>From the article: "The record business wants to lock down songs on a new breed of CDs. Is this the end of rip, mix and burn? We visit the labels' laboratories to learn more about what 'stickiness' means to music." Read the full article here: http://www.inside.com/jcs/Story?article_id=26900&pod_id=9 Oh, and CDparanoia is mentioned in the article :-) Greetings,

> Hi glad you're interested - comments below. > > On Fri, Jan 15, 2010 at 5:05 PM, Brian Willoughby <brianw at sounds.wa.com > > wrote: > > On Jan 14, 2010, at 20:59, Nicholas Bower wrote: > I'm after the cheapest way to decode a flac stream with following > criteria; > - Transport from UPnP (DNLA) NAS using either WiFi or Cat5 networking > - Toslink