similar to: Ask for suggestions about optimizing opus on STM32F407

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);

Hello Forrest, some years ago i developed a network media player based on a STM32F407ZGT6 (168MHz clock) and opus 1.1. I used just the fixed point code and did no particular optimization on the opus code itself because the performance was already quite good, see figures below. The figures are for real time playback with different frame sizes and various constant bit rates. I didn't play

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

Hey, I am having trouble with the size of the encoded bytes by Opus. I am also having issue with the Bandwidth ctl. Here is the scenario. If I encode 16khz sampled audio: opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND)) opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&x)) = 1102 opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&x)) = 1103 average encoded size = 120 bytes if I

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:

Hi everyone, I?m currently trying to use opus on a ST ARM (STM32F407) without any OS (bare metal). The aim of my project is to transmit voice over CAN bus. The main issue I have is that opus fail to allocate memory, the ALLOC macro always return a NULL pointer. I have sure that I have enough free space to allocate buffers. Is there anyone who already try this or have meet this issue ? Thanks

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

HI, I'm trying to simulate an audio conference where each leg can be with a different bit rate. This needs to encode the source PCM to to different bit rates back to back and store and send respective encoded frames/packet to the respective channel. For this I changed the opus_demo as below. But the output of the second encoded frames is completely garbled. Appreciate if anyone can suggest

Here is a simple patch to fix a bug in opusenc's set-ctl-int code -- Thanks, Michael Graczyk -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fix-potential-uninitialized-access-for-set-ctl-int.patch Type: text/x-patch Size: 992 bytes Desc: not available URL: <http://lists.xiph.org/pipermail/opus/attachments/20160425/22994ffa/attachment.bin>

If you encode with int iByteSizeEncoded = opus_encode(m_enc, m_ShortsInput, (1920/sizeof(short)), m_EncodedBytes, (960*6)); ... the byte size is 120. I thought that an odd value like 359 has to be a fault on my side. I thought it would be a multiple of something. So 359 is really correct??? Thank you for clearing this up. Am 06.10.2013 16:50, schrieb Jean-Marc Valin: > Why

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

Hi! Short version of my question: How to produce Opus frames which can be safely concatenated and how to embed them into a WebM file? Long version: I'm currently implementing a web-based audio player which streams audio as opus/WebM using the HTML5 media source extensions. Currently, the server decodes a set of input files to a fixed RAW audio format (stereo, 48000 kHz) and encodes the

Hi everyone, Some time ago, I sent a pull request <https://github.com/xiph/opus/pull/107> to the Opus github page. Jean-Marc asked me to post it to the mailing list so everyone can have a look at it. You can find the description and code changes below. Please let me know if you have any questions or concerns. Best regards Gustaf Ullberg In WebRTC, we would like to be able to

Hi All. I am seeing the following unexpected behavior with OPUS_SET_MAX_BANDWIDTH. I expect that setting this to OPUS_BANDWIDTH_NARROWBAND would give similar results to passing an 8Khz sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl

Hi, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

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

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

Hi Jean-Mark, thank you for your answer! Yes, you understood my question correctly. I was just about to compose a reply to my original question, where I described how I solved my problem. As you've already suggested, I've switched to Ogg/Opus, which is better supported, but does not work with the Media Source Extensions. I'll have a look whether disabling prediction will help with

Thank you Mark. I agree and have now updated the pull request with a new commit, addressing your comments. Please take a look. /Gustaf On Fri, 5 Apr 2019 at 11:41, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-01 3:37, Gustaf Ullberg wrote: > > Hi everyone, > > > > Some time ago, I sent a pull request > > <https://github.com/xiph/opus/pull/107>