similar to: opus_demo produces garbage

Dear Christian van Bijleveld, You can use any of the below DSPs of Texas Instruments 1. TMS320C674x - This supports floating point implementation of opus 2. TMS320C66x - This supports both floating and fixed point implementations 3. TMS320C64x - This supports only fixed point implementation Regards, Mahantesh On Mon, May 13, 2013 at 10:12 PM, <opus-request at xiph.org> wrote: >

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,

Hi I used opus_demo to clarify how opus works, I try this command: opus_demo -e" audio 8000 1 8000 -cbr /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus And I get the file test.opus, then comes the first problem, this file hasn't header so i can't reproduce it using an external tool, how can i add a header to this file Then when i trie to run opus_demo -d 8000 1

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

Hi, I have downloaded the latest stable version *1.0.2 opus audio codec* from the following link: http://www.opus-codec.org/downloads/ *Details of OS & CPU:* OS : Microsoft Windows XP CPU : intel core 2 Duo cpu I am using Microsoft visual C++ 2010 Express to build the codec (opus.vcxproj available in the package downloaded). I am able to build it & could generate the application

--- src/opus_demo.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/src/opus_demo.c b/src/opus_demo.c index 72506b5..5a75679 100644 --- a/src/opus_demo.c +++ b/src/opus_demo.c @@ -245,14 +245,14 @@ int main(int argc, char *argv[]) double bits=0.0, bits_max=0.0, bits_act=0.0, bits2=0.0, nrg; double tot_samples=0; opus_uint64 tot_in, tot_out; - int

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

On 2013-10-21 5:08 AM, Antonio Juan wrote: > I used opus_demo to clarify how opus works, I try this command: > > opus_demo -e" audio 8000 1 8000 -cbr > /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus opus_demo doesn't support .opus files. It uses a special raw format just for testing. You want opusenc and opusdec from the opus-tools package. -r

On 21 November 2014 at 18:06, Timothy B. Terriberry <tterribe at xiph.org> wrote: > > Viswanath Puttagunta wrote: >> >> a. Simplest use case to validate this optimization for correctness. >> b. Simplest use case to validate this optimization for performance. >> >> Would prefer something like opusdec that can be executed on command >> line. > >

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

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

I didn't expect this, is it normal? When you use inbandfec, and have packet losses, you end up with more audio samples than you started with. With short frame (2.5), the FEC isn't supposed to do anything but it shouldn't do that should it? and 20mS frame, should do 'something' but? add audio? I'm using opus_demo for this. Here's the steps (I have my test audio in

Hello OPUS, It appears to me that multi-frame packets are not supported by opus_demo.c (e.g. 30ms packest containing 3 10ms frames), can you confirm. However RF6716 does support such packets. Thanks Cliff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20131121/56bab79a/attachment.htm

>> >> a. Simplest use case to validate this optimization for correctness. >> >> b. Simplest use case to validate this optimization for performance. >> >> >> >> Would prefer something like opusdec that can be executed on command >> >> line. >> > >> > >> > The easiest thing to use is probably opus_demo (opusdec

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

I wonder if anybody try to compress a pure sine wave using OPUS codec. When I compressed the mono 1KHz, 16bits 48000 samples per sec. audio stream using the 'opus_demo' utility: opus_demo -e audio 48000 1 2min_1kHz_Sine_16bit_48kHz.wave 2min_1kHz_Sine_16bit_48kHz.opus_raw I had the output stream that is shown below. 00 00 00 01 00 00 00 00 78 00 00 00 01 00 00 00 00 78 00 00 00 01 00

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

Hi, I am interested in this part of the answer: "Make sure both are also linked to the same libopus (opusenc could be using the system libopus 1.0.x while opus_demo is from git)" I dowloaded a tarball distrib of libopus (1.1-beta),build and installed it,and afterwards did the same with opus-tools (0.1.7),using the former installed libopus library (also using libogg 1.3.1). When I execute

I suspect the main difference is due to time alignment of the samples and the exact trimming at the beginning and at the end of the encoded files. I also note that there's a missing "--" in front of "bitrate" and only one "-" instead of two for cvbr in your opusenc command line. Make sure both are also linked to the same libopus (opusenc could be using the system

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