similar to: Developing OPUS on TI CC3220

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 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

On 02/20/2018 06:48 PM, Rodriguez, Vince wrote: > 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. Not sure what you mean here. If your compiler supports C99 VLAs, then by all means just define VAR_ARRAYS and you're good to go. If not, the second best option is alloca(),

Hi Vince, Normally, Opus should work fine on the CC3220, though real-time encoding may be a bit tight depending on what configuration you choose. You'll probably want to configure with --enable-fixed-point and --disable-float-api (or define FIXED_POINT and DISABLE_FLOAT_API if you don't use autoconf). The encoder has a bunch of run-time settings that you may want to set, depending on

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, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Hello, for your consideration. The following patch moves the channel_pos() function from within the #if !defined(DISABLE_FLOAT_API). This change is required when compiling with FIXED_POINT and DISABLE_FLOAT_API defined. #### ### diff --git a/src/opus_multistream_encoder.c b/src/opus_multistream_encoder.c index 3efab53..6f3eb53 100644 --- a/src/opus_multistream_encoder.c +++

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 everyone, Just to let you know that version 1.2beta3 is out. The main change is the split into libspeex and libspeexdsp. Basically, libspeex keeps the codec (just like it was for 1.0.x) and libspeexdsp has all the new components. Other changes include a new jitter buffer algorithm and resampler improvements/fixes. This is also the first release where libspeex can be built without any floating

hi, I am porting opus encoder to a low end device with 32K ram, 256K flash and 32MHz arm M3 mcu. But opus seems consume too much. To make it work , what I can think of 1, Only fixed point supported 2, Only mono voice application supported 3, Set complexity to zero 4, Support only one sample rate, like 16KHz 5, Silk mode only or Celt mode only My question is , before

Hi to everyone, this is my first post to the list. I'm developing an application for a Fujitsu MCU (90364 series) using Fujitsu Softune6 Workbench in linux under wine. Everithing works well exept for the firmware serial uploader. It starts and sends some commands to the MCU but hangs with a timeout error (obviously I've correctly simlinked and chmodded the appropriate ttyS*). To

In fact, on my shelf is running speex using Raisonance Primer2 DK and IAR STM32-SK DK. But I want to save the power consumption of the device runnig speex, and EFM32 seems to be a very good candidate. Also, do yo think that the STM32F103 (Cortex M3 90 MIPS) could be used to run Speex at 4 Kbps in real time? Regards, Nicolas 2010/3/3 Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> >

Hi, I am working on trying to compile opus to be as small as possible. I have passed the FIXED_POINT and the DISABLE_FLOAT_API but I am looking into making this lib even smaller in size. Where can I find out about how to best strip this lib to only compile in what I need?

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);

I've been looking closely at LLVM as a means to developing a new toolchain for an MCU core of very similar architecture. To that end, the once included PIC16 backend might be a valuable reference. I found a message in April of this year that indicated it had been dropped from new releases however, and that were it to be resumed "it will be largely a rewrite". I'm wondering if

Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came

Hi, I've attached a patch to fix use_dtx when building with DISABLE_FLOAT_API. Thanks, Felicia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20160720/4bcbab8d/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fix-use_dtx-for-DISABLE_FLOAT_API.patch

Hi. I found that DTX is always unavailable when DISABLE_FLAOT_API is not defined in OPUS_AUTO mode. As you know analysis_info.valid is alway true except NaN case and is_silence is alway true except digital zero signal case. In general, following condition will be alway true except exceptional case. (analysis_info.valid || is_silence) But in a code, there is a NOT expression in front of above

>> Do you have a link? Google isn't turning this up. >http://lists.cs.uiuc.edu/pipermail/llvm-commits/Week-of-Mon-20080512/062319.html Wow, PIC looks like a brutal port. AVR would be easy in comparison: stack-based architecture with plenty of registers. However, these architectures share: - 8-bit loads, stores, and arithmetic ops. - Harvard architecture What sizes and

Hi, Do you think that Speex could run in real time on the new EFM32 ultra-low consumption Cortex-M3 MCU (180 ?A/MHz) (NB 8KHz - Quality 4 - complexity 1 / same as STM32F103 Speex lib.) ? Its max speed is 32 MHz (about 40 MIPs). Thanks. Regards, Nicolas -------------- next part -------------- An HTML attachment was scrubbed... URL: