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 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 all ! If someone could give me a hint on how to proceed with the following i'd be very happy: I have a test setup on an nrf52832 (Cortex M4) in which I receive audio from a PDM microphone (64 sample frame) and pass it directly to an I2S device i.e. from ISR to ISR. With uncompressed audio this works just fine. Now I try to insert OPUS1.3 in the path but cannot make it work. The

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()),

Hello All I have successfully run the Opus Decoder for 16-bit WAV files. However when doing the same on 8-bit, the decoder produces noise, but on 16 bit data the output is working. Both the 8 and 16 bit files are from the same source and hence except for some loss of quality on 8 bit, they are identical in total play back duration. For both 8 and 16 bit data I have used the following parameters

Hello Ralph, > Likewise opus_encode() takes 16 bit samples, so you need to extend each > sample from an 8 bit source before encoding. Two questions 1. In opusenc.c which API does the extending the 8-bit to 16-bit? 2. If that is the case then how will 24 bit PCM sample work? Regards Amit On Thu, Jan 7, 2016 at 12:21 PM, Ralph Giles <giles at thaumas.net> wrote: > On 07/01/16

---------- Forwarded message --------- From: Peter Svensson <petersvenss85 at gmail.com> Date: tis 19 feb. 2019 kl 20:43 Subject: Re: [opus] Custom mode To: Emily Bowman Hi Emily ! Thank you for responding. I think my problem is not (yet) with OPUS itself. Encoding at complexity 0 takes 1.6ms ( 4.342ms at complexity 10 !) and decoding takes 1.9ms. 3.5ms, out of my 4.096ms budget, is

Hi, I got libopus 1.2 from the download page. I compiled it using visual studio 2015 with your configuration (Release). I integrated opus.lib and the new include files in my own solution, but when I compile, I found 28 link errors (lnk 2001): - silk_Encode - ec_enc_init - celt_inner_prod_sse - opus_select_arch - silk_InitEncoder - ec_enc_shrink - silk_log2lin - ec_enc_bit_logp -

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

Hi Opus list, Please forgive me if this has been asked before. I find that Opus encoder created in mode OPUS_APPLICATION_AUDIO (as opposed to _VOIP) is using a lot of bits to encode silent periods of speech. This is relevant to a voip application for which good quality music is desirable, and in which I add a minimal comfort noise (order of few bits loud, e.g. MLS signal of amplitude 1 or 2)

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

Jean-Marc, Thanks for the response and the helpful info. I am trying to get the library to build without using the pseudostack define, and use either VAR_ARRAYS or ALLOC, but it seems the global stack is not defined. Where do can I define this in my example? VR -----Original Message----- From: Jean-Marc Valin [mailto:jmvalin at jmvalin.ca] Sent: Tuesday, February 20, 2018 5:40 PM To:

On 07/01/16 10:04 AM, Amit Ashara wrote: > opus_decoder_ctl(sOpusDec, OPUS_SET_LSB_DEPTH(ui32BitsPerSample)); OPUS_SET_LSB_DEPTH only affects the encoder. If you check the return value here you should get OPUS_UNIMPLEMENTED. > output_samples = opus_decode(sOpusDec, (const unsigned char > *)&pcRdBuf[0], len, opi16_out, (ui32SizeOfWrBuf/ui8ScaleFactor), 0); I suspect the issue is

Hello Benjamin, The original WAV file I have is linear 8-bit PCM. I want to ensure that original linear formats are kept as is. Later I will add support for ulaw. Regards Amit On Fri, Jan 8, 2016 at 5:34 PM, Benjamin Schwartz < benjamin.m.schwartz at gmail.com> wrote: > Do you really need linear 8-bit PCM or do you want ulaw? Linear 8-bit is > ... pretty rare. > > On Thu,

<style>table.customTableClassName {margin-bottom: 10px;border-collapse: collapse;display: table;}.customTableClassName td, .customTableClassName th {border: 1px solid #ddd;}</style><div id="write-custom-write" tabindex="0" style="font-size: 12px; font-family: 宋体; outline: medium none currentcolor;"><p style="margin:0px;">Dear

Because a wideband signal has both a narrowband part and a wideband part the first part of the wideband decode is to call the narrowband decode. In the fixed point version the signal generated from the narrowband decode is downshifted at the end. In this same version the wideband then upshifts the signal by the same amount. I would like to do away with the downshift on the narrowband side

Is there a way to tell during the NB decode if it was kicked off by the SB decode? I would like to avoid saturating and packing the output of the NB decode, only to unpack it for the SB. Thanks. -Fritz

I made yet another new wiki page documenting the mail configuration file. Looks really nice. We can all start working on improving the docs in this file and eventually merge the docs here back into the example config file. http://wiki.dovecot.org/moin.cgi/MainConfig -- Marc Perkel - marc at perkel.com Spam Filter: http://www.junkemailfilter.com My Blog: http://marc.perkel.com

I wrote a proof of concept regarding the cpu capabilities runtime detection and choice of optimized function. I follow design which had been discussed on IRC. Also, i notice a little drawback: we must propagate the arch index through functions which don't have codec state as argument. However, if it's look good, i will continue to implement it. Best regards, -- Aur?lien Zanelli