search for: spdif

Hi, i want send sound to my yamaha amplifier trough digital output from my audio-card. When I make: speaker-test -Dplughw:0,1 -c2 i can hear the sound. If I set the DevicePCM1 to default i can hear sounds from analog output trying to set hw:0 hw:0,1 plughw:0,1 didn't produced any sound (analog nor digital). How should I setup Alsa Driver in the wine registry? asound.conf and .asoundrc is

...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 separate from digital circuits. This would seem to point to SPDIF and AES3 as the ideal interconnect, but the unfortunate truth is that those two specific digital audio standards create more problems than they solve. I'll try to explain: Every digital audio playback system combines some sort of storage media with some sort of DAC. Between these basi...

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 generate assigned clock signal. But if playin...

...do to use AEC in most computers? (Don't tell me to change a sound card.) >> >> From: Steve Underwood <steveu at coppice.org <mailto: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? >> >I already said they do, and what the lock to - SPDIF ports. Most...

...can playback 48kHz music. This is > because only few of them have bought their soundcards during the last > 1-2 years when 48kHz support became common. Only the digital I/Os on > some soundcards have to resample in order to support 44.1kHz ... there > was some ranting about the very bad SPDIF of the SB Live!, but I can't > recall what the exact problem was, but I think it was something about > the resampling and 44.1 vs 48 kHz. SB Live! and Audigy resamples EVERYTHING to 48 kHz, even if you just connect a CD-ROM with SPDIF and send it to the SPDIF output. AFAIK :-) -- Dav...

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 ?" Bes...

...it a/arch/arm/boot/dts/stih407-family.dtsi b/arch/arm/boot/dts/stih407-family.dtsi > index d263c96..bdddf2c 100644 > --- a/arch/arm/boot/dts/stih407-family.dtsi > +++ b/arch/arm/boot/dts/stih407-family.dtsi > @@ -956,5 +956,31 @@ > st,version = <5>; > st,mode = "SPDIF"; > }; > + > + sti_uni_reader0: sti-uni-reader at 0 { > + compatible = "st,sti-uni-reader"; > + status = "disabled"; I find it's normally nicer to place the status of the node at the bottom, separated by a '\n'. There isn't a functio...

...it a/arch/arm/boot/dts/stih407-family.dtsi b/arch/arm/boot/dts/stih407-family.dtsi > index d263c96..bdddf2c 100644 > --- a/arch/arm/boot/dts/stih407-family.dtsi > +++ b/arch/arm/boot/dts/stih407-family.dtsi > @@ -956,5 +956,31 @@ > st,version = <5>; > st,mode = "SPDIF"; > }; > + > + sti_uni_reader0: sti-uni-reader at 0 { > + compatible = "st,sti-uni-reader"; > + status = "disabled"; I find it's normally nicer to place the status of the node at the bottom, separated by a '\n'. There isn't a functio...

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 ?" Bes...

>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,

...seen are either expensive boutique heavy-weights or portable > devices. > There are certain some over-priced boutique devices out there, but > I'd say that much of the price goes into working around the > deficiencies in unidirectional digital audio connections such as > SPDIF and AES3. The real problem is that there is not a better > interconnect for HiFi separates other than FireWire and USB, which > each require a level of complexity in the device which is more > difficult for the average person than pure analog or traditional > digital I/O. >...

..."; > + }; > + > + sti_uni_player3: sti-uni-player at 3 { > + compatible = "st,sti-uni-player"; > + status = "disabled"; > + #sound-dai-cells = <0>; > + st,syscfg = <&syscfg_core>; > + clocks = <&clk_s_d0_flexgen CLK_SPDIFF>; > + assigned-clocks = <&clk_s_d0_quadfs 3>, <&clk_s_d0_flexgen CLK_SPDIFF>; > + assigned-clock-parents = <0>, <&clk_s_d0_quadfs 3>; > + assigned-clock-rates = <50000000>; > + reg = <0x8D85000 0x158>; > + interrupts = &lt...

..."; > + }; > + > + sti_uni_player3: sti-uni-player at 3 { > + compatible = "st,sti-uni-player"; > + status = "disabled"; > + #sound-dai-cells = <0>; > + st,syscfg = <&syscfg_core>; > + clocks = <&clk_s_d0_flexgen CLK_SPDIFF>; > + assigned-clocks = <&clk_s_d0_quadfs 3>, <&clk_s_d0_flexgen CLK_SPDIFF>; > + assigned-clock-parents = <0>, <&clk_s_d0_quadfs 3>; > + assigned-clock-rates = <50000000>; > + reg = <0x8D85000 0x158>; > + interrupts = &lt...

I have a GeForce G 105M. That's, I guess, an NV50 family video card: https://nouveau.freedesktop.org/wiki/CodeNames/#NV50 My question is... does Nouveau support HDMI with this video card? According to https://nouveau.freedesktop.org/wiki/FeatureMatrix/ TVout is "TODO" and HDMI audio is "DONE". What about HDMI video? Is that grouped under TVout or is TVout supposed to be

...sti_uni_player2: sti-uni-player at 2 { + status = "okay"; + }; + + sti_uni_player3: sti-uni-player at 3 { + status = "okay"; + }; + + sti_sasg_codec: sti-sasg-codec { + status = "okay"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdif_out>; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "sti audio card"; + status = "okay"; + + simple-audio-card,dai-link at 0 { + /* DAC */ + format = "i2s"; + mclk-fs = <256>; + cpu { + soun...

...ed, 26 insertions(+) diff --git a/arch/arm/boot/dts/stih407-family.dtsi b/arch/arm/boot/dts/stih407-family.dtsi index d263c96..bdddf2c 100644 --- a/arch/arm/boot/dts/stih407-family.dtsi +++ b/arch/arm/boot/dts/stih407-family.dtsi @@ -956,5 +956,31 @@ st,version = <5>; st,mode = "SPDIF"; }; + + sti_uni_reader0: sti-uni-reader at 0 { + compatible = "st,sti-uni-reader"; + status = "disabled"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + reg = <0x8D83000 0x158>; + interrupts = <GIC_SPI 87 IRQ_TYPE_NONE&gt...

I have post a message about this topic @questions but I have not get feed back. My laptop has an ALi M5451 Sound Card. I am using FreeBSD4.8-STABLE with the "device pcm" activated in the kernel. At boot time I get this output from the kernel: pcm0: <Acer Labs M5451> at device 4.0 on pci0 pcm0: unable to map register space device_probe_and_attach: pcm0 attach returned 6 Searching

...the dynamic range (and then some). I *think* the opusenc command-line tool can handle 24-bit wav files, but I could be wrong (if not it's something that needs to be added). Cheers, Jean-Marc On 06/23/2015 11:04 AM, Kazem Baadpie wrote: > 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 samp...

...-family.dtsi b/arch/arm/boot/dts/stih407-family.dtsi > > index d263c96..bdddf2c 100644 > > --- a/arch/arm/boot/dts/stih407-family.dtsi > > +++ b/arch/arm/boot/dts/stih407-family.dtsi > > @@ -956,5 +956,31 @@ > > st,version = <5>; > > st,mode = "SPDIF"; > > }; > > + > > + sti_uni_reader0: sti-uni-reader at 0 { > > + compatible = "st,sti-uni-reader"; > > + status = "disabled"; > > I find it's normally nicer to place the status of the node at the > bottom, separated by a...