similar to: Force Opus/CELT to encode 2 mono instead of 1 stereo

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

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

Hi Jean-Marc, On Jun 15, 2013, at 12:20 PMEDT, Jean-Marc Valin wrote: > > >> So I still wonder, if you set up a custom mode, but then had all the >> settings the same as a normal mode, would the codec perform worse, or >> the same? > > You'll have to try normal vs custom modes and choose. The only thing I'm > telling you is don't run a 48 kHz

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).

Is there a configuration or compile flag that lets me disable the SILK portion of the codec and use CELT only? I could have sworn that there is something, but I can't seem to find it in the mailing list archives. The application here is that I am attempting to update from the old CELT codec to OPUS. Unfortunately, the CELT codec was running *very* close to the CPU (MIPS32--80MHz) limit

Is there a particular reason why CELT grabs 100KB of stack immediately? Is that really required or can that be trimmed down some/a lot? -a

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:

Thanks for the reply. For my application I unfortunately need a better signal reconstruction. It's not necessarily a problem that the constant DC voltage is removed, but the audio signal will need to be more exact because it's also studied visually. To be more exact, I need to record and transmit audio data recorded from animals (mostly birds). Are there ways to achieve a more similar

I need to test the latency of a system that I ported CELT to. I'd like to set a flag when the stream "changes". In order to see the stream "change", I'd like to be able to scan the compressed CELT directly to set the flag. ie. I'd like to be able to find a byte or two that change reliably between two values when the stream "changes". I don't

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);

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,

Hi, Does Celt (in particular version 0.7.1) exploits correlation between the two channels in stereo mode? In practice, is it possible to use the two channels as two mono signals without affecting quality? Thanks Best Regards Riccardo Riccardo Micci Senior DSP Engineer, Wireless Group Cambridge Consultants Science Park, Milton Road Cambridge, CB4 0DW, England Switchboard: +44 (0)1223

Hi, If I want to encode two audio channels that are not related, audio from two different sources not stereo, is it best to create two instances of a single channel encoder or use one encoder with two channels? thank you, Paul.

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

Hello, I'm not a programmer or Opus developer but I tried to test the sound quality of a music file (Nick Warren - Devil's Elbow, freely available from the author's Soundcloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary 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

I have been trying some different sample rate and bitrate combinations to get a feel for how CELT behaves. I then encoded it with a couple of different frame sizes and sample rates. There were some small differences, but nothing horrible. So, I decided to run with 24KHz sample rate with 16 bit samples. This seemed like a reasonable tradeoff against the quality degradation that my folks

Marc Lindahl wrote: > Of course, I'm setting up a bunch of tests to evaluate these, what I was asking was more along the lines of, > If you set up the same exact, including the sample rate, do you get the same results (e.g. same code path, calculations, etc.?) If you configure a custom mode with the standard parameters (48 kHz sampling rate and a frame size of 120, 240, 480, or 960),

All, Just a heads up that the IETF codec working group has issued a last call on the draft Opus specification. This means we think the draft adequately documents the format, and that we're willing to live with whatever bugs are present in the reference implementation, if they can't be fixed without breaking decodes. In turn, this means we could really use some feedback from implementers.

Hello Jean-Marc, yes, it works for both 32 and 48 kb/s. Also this option marginally increases the size of the file which is understandable. I tried a few other music files and found that this stereo narrowing happens not only in the case of track I submitted (although in smaller scale) so it would be great if the final libopus 1.2 had officialy a parameter to force music "mode" of