similar to: DSPs which are suitable for porting OPUS

Hello All, We have been doing an optimised port of OPUS to a ARM Cortex A9. We are currently measuring between 20 and 90 MCPS for our code running on a Panda board (single core), this covers all bit-rates,sample rates for stereo coding (encode + decode) under normal operation. As Marc says complexity can be controlled via the API with our higher figure corresponding to the default setting of

Hello! I encoded a voice file (48kHz) with opusbin\opusenc.exe with the standard settings and decoded it. The output was amazing. I could not hear any loss at all. Then i encoded the same file with opus_demo.exe and standard settings and then decoded it. The output had a sizzling noise, even when I used full bandwidth. I think I have played around with any of the settings in opus_demo.exe,

Hello, I am interested in porting OPUS to an embedded platform. The idea is that the encoder and decoder will run on different processors. In order to choose the proper platform, I need an estimate of the resources which are needed for OPUS. I read the following in the OPUS wiki: "The Opus code base is written in C89 and should run on the vast majority of recent (and not so recent) CPUs.

The e-mail sent to you contained an attachment with a not allowed filetype. Please inform the sender to pack this type of attachment into a password protected ZIP-archive. Eine an Sie gesendete E-Mail enthielt einen nicht erlaubten Dateianhang. Bitte informieren Sie den Absender, diese Art von Anhang kann nur als Passwort gesch?tztes ZIP-Archiv versendet werden. Details: Sender:

HI Hermann, I am not sure I understand exactly what you are trying to do. Let me try to describe what I understood (I assumed your byte array starts @ 1 and not 0 as it is supposed to be in C) : ############################## CASE 1: (output is good) ############################## - INITILIAZE_DECODER (reset state memory) - READ bytes 1 to 124 - SpxDecode(DinBuf, DoutBuf, unsigned int

I'm not using Opus at all. I'm just including its resampler in my own sources. It's not even a DLL; it's directly compiled together with the rest of my code. You need these sources from the opus-tools package (http://www.opus-codec.org/downloads/): arch.h resample.c resample_sse.h speex_resampler.h stack_alloc.h In your project file, define these macros: #define

My file is 3 hours long, so decoding takes around 5 minutes on an average computer. That is a bit too long unfortunately... Am 23.12.2011 20:38, schrieb Steve Checkoway: > On Dec 23, 2011, at 10:54, Hermann Weber<hermie.weber at gmx.de> wrote: > >> And how many frames does Speex need to "recover"? >> Or is that not predictable? > No idea. My guess is not

When I already decoded something, and the decstate is not erased or reset, shouldn't that be sufficient to decode other frames with a good quality? For example, when I have decoded 30 frames without decoding the previous frames, the sound quality may not be too good, but afterwards the decoder should (in my opinion) be in a "good" state. But when I decode other frames

Dear Nikos, thanks! But you use Opus only for resampling, not for entirely replacing Speex, don't you? Greetings! Hermie Am 07.05.2013 22:53, schrieb Nikos Chantziaras: > The Opus resampler is actually a bugfixed version of the Speex one. Same > interface/API, but with the bugs removed. It's why I recommended it :-) > Otherwise I would have recommended something entirely

The Opus resampler is actually a bugfixed version of the Speex one. Same interface/API, but with the bugs removed. It's why I recommended it :-) Otherwise I would have recommended something entirely different, like SoXR. On 07/05/13 22:12, Hermann Weber wrote: > Dear Nikos, > > thank you very much for your advice! > > I am not a little bit new to C++, and experiments

Sorry, it seems I have only replied to Lakhdar, not to the newsgroup. Below is my reply to Lakhdar, and I would like to make it more clear now, using some pseudo values for simplicity: I read bytes 1 to 124 from my encoded spx file. I decode themt and get the values: ---Frame 1---- -293 -8234 2134 17 ---Frame 2---- -9323 -732 189 2329 Both frames are just perfect as I need them. But now when I

Dear Nikos, thank you very much for your advice! I am not a little bit new to C++, and experiments (like using another third party library) always turn out to be days of work for me. That is why I would first like to try Speex's new internal resampling function. I am currently still using the old 1.0.4 version. I have now downloaded the new 1.2rc1 version. Unfortunately it does not want

And how many frames does Speex need to "recover"? Or is that not predictable? Greetings, Hermann Am 23.12.2011 19:17, schrieb Steve Checkoway: > > On Dec 23, 2011, at 10:03 , Hermann Weber wrote: > >> I found out something: >> >> The more frames I decode before the frame that I actually want to >> decode, the better the quality becomes. >> For

Hello, I am testing the command line based opus encoder/decoder tools and I have some questions regarding the information given by the encoder (see below as an example). Encoding using libopus 1.0.2 (audio) ----------------------------------------------------- Input: 48kHz 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 64kbit/sec VBR Preskip: 312 Encoding complete

Hello! I am running opus_demo.exe with the following cmds: -e audio 48000 1 16 -cbr m:\test\test1.raw m:\test\test1.opus When I open the .opus file with a hex editor, I see only

I found out something: The more frames I decode before the frame that I actually want to decode, the better the quality becomes. For example when I basically want to decode frame #100, I read frame #80 to #100, and then frame 100 has the quality that I need. Why? Is there any information on this behaviour? Thank you. Hermann Am 23.12.2011 18:37, schrieb Hermann Weber: > It would be nice

Hello, I just started into the world of OPUS. I would like to know what is the best way to build the Opus library and Opus tools on a Windows 7 pc. I see that the Opus library and the Opus tools packages have a few Makefiles. It also mentions mingw as an option for building. Is it somewhere explained what are the steps needed, or does anyone have some tips? Thanks, Met vriendelijke

Hello! I would like to know if it is possible to request encoding to a higher samplerate than the original data. For example my uncompressed data was 22 kHz, and I would now like to decode the compressed data to 48 kHz (to feed it to a filter which expects a high "input" format). Is this possible with Speex? Thank you. Hermie

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! I need to extract audio data at a certain position in respect to the original audio data. Is the "-cbr" switch meant to ensure that data can be found on a specific position? I want to be able to predict where certain audio can be found in the encoded data. An example: If you put 10 apples into a box and then compress the box, you will not be able to predict where apple C