similar to: Issue with decoding 8-bit PCM data

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

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,

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

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

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:

<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

Hello Mark, The resulting 8 bit file has a lot of squelching noise compared to the 16 bit output from OPUS decoder. During encode I am using popi16fmtBuffer[ui32Loop] = (opus_int16)pcRdBuf[ui32Loop]; And during decode since the data is in the lower 8 bit I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara

On 02/23/2018 10:47 AM, Rodriguez, Vince wrote: > Currently decode takes 16~22 ms and encode is ~13 ms. What is the best > way to try to reduce this time? Also, unsure why encode is taking less > time than decode... Tried running a profiler? > I've also noticed if I change my encoder to silk only, I get a Heap > overflow... I have dynamically allocated a heap for the codec,

Hello All, Turned out to be a coding error. The modified buffer was not being allocated to write back to the File System. After correction the mono 8 and 16 bit for 8K-48K works well now. Regards Amit On Mon, Jan 11, 2016 at 11:05 AM, Amit Ashara <ashara.amit at gmail.com> wrote: > Hello Mark, > > The resulting 8 bit file has a lot of squelching noise compared to the 16 > bit

opus_decode() produces 16-bit signed linear PCM, and opus_decode_float() produces 32-bit floating point PCM that is useful when you want a higher bit depth. If you need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed integers for 8-bit linear PCM. (It uses signed for

We have found that it is possible to achieve a 30 to 50% reduction in MHz requirement for implementation of OPUS on Cortex M4 compared to the public version (v1.3 beta/v1.2.1). For the CELT configuration you mention (complexity 0, 16kHz, mono, 20ms) we are measuring a 4ms encode time and a 3ms decode time for that platform (32kbit/s). An important issue that I haven't seen much discussion