similar to: 24 bits samples

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

Hi, Opus itself supports 24-bit audio and can handle all of 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

Hi It pretty works for 16 bits samples but not for 24 bits samples. In fact, I am going to implement opus on ADSP-21489 (DSP processor from Analog Devices), I compiled "celt directory" from opus 1.1 and used "opus_custom_demo.c". It hasn't any problem with 16 bits samples but when I use 24 bits samples , there is no output audio. opus_decoder will still generate 16 bits

I finally could download and install the swirl package but when I want to load it using library (swirl) it gives the following message package ?swirl? successfully unpacked and MD5 sums checked The downloaded binary packages are in C:\Users\Emdad notebook\AppData\Local\Temp\RtmpEJyZTB\downloaded_packages > library(swirl) Error in loadNamespace(j <- i[[1L]], c(lib.loc, .libPaths()),

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

>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,

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

I saw the custom API, but nothing explicitly says "CELT-only" just "custom sample rate and frame size". I'll dig further now that you've pointed me in a direction. Thanks, -a On 7/6/15, 6:18 PM, Jean-Marc Valin wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > I believe what you want is called Opus custom (OPUS_CUSTOM in the > code).

Hello All, This is my first appearance to the opus mailing list. My name is Michael Mehari and i am a researcher at Ghent University mainly on cognitive radio networking. Recently i am working on a demo scenario on self adjustable wireless audio conferencing and i selected opus for the audio encoding and decoding part. specifically the applications (i.e opusenc and opusdec) from opus-tools

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

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

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

On Fri, 26 Aug 2016, Peter Griffin wrote: > This patch adds the DT nodes for the uniperif player > IP blocks 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 | 76 +++++++++++++++++++++++++++++++++++ > 1

On Fri, 26 Aug 2016, Peter Griffin wrote: > This patch adds the DT nodes for the uniperif player > IP blocks 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 | 76 +++++++++++++++++++++++++++++++++++ > 1

https://bugs.freedesktop.org/show_bug.cgi?id=62305 Priority: medium Bug ID: 62305 Assignee: nouveau at lists.freedesktop.org Summary: No HDMI Audio out (pass-thru) on Geforce 7600 Go QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: doug at

Hi, I have a question on the audio data format that your librarie encodes. Must the audio data be in a planar format or interleaved format ? Thanks in advance, Adrien -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170404/2c2c871e/attachment.html>

Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8" jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien