similar to: Exact audio position

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: >

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

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

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

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

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

I just had two quick questions about opus. When using the encoder, how do I use a live microphone as the source, and how do I stream the output to a server? (Would it work with rtp, http, etc.?) Thanks! :) -- <>< Greg Bernard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20130807/5e6e57bd/attachment.htm

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 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

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

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

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

Hello! I was using Speex all the time, and I am now moving to Opus. I had encapsulated the decoder a bit, I had the following cpp file: #include "StdAfx.h" #include "spxcodec.h" #define MAX_FRAME_SIZE 2000 #define MAX_FRAME_BYTES 2000 CSpxCodec::CSpxCodec() : enh(1), rate(8000) { } CSpxCodec::~CSpxCodec() { } void CSpxCodec::Init() { speex_bits_init(&bits);

On 12-Feb-08, at 2:54 PM, Silvia Pfeiffer wrote: > I've added Monty's email to the wiki at http://wiki.xiph.org/ > index.php/GranulePosAndSeeking , but I was unable to edit the wiki > entry page and add a "Granulepos and Seeking" link under the > "developer resources" section. Maybe somebody with more rights on > the wiki could add this. I'm

Hello, I'm currently using Liquidsoap to stream a group of files to Icecast. The stream is sent to Icecast in a few different formats (MP3,HE-AAC v2,OggFlac,OggOpus). The "Current Song" metadata is correctly displayed for the MP3 and the AAC stream, but not the OggFLAC or OggOpus stream. I see a few entries in the Icecast forums, but it does look like FLAC metadata support never

On 13/02/2008, Ralph Giles <giles@xiph.org> wrote: > > The hypothetical seeking algorithm. We should be writing it down, I > suppose This is about the closest we have to a source document on that: http://web.archive.org/web/20031201054855/http://www.xiph.org/archives/theora-dev/200209/0040.html K.

I have almost finished integration of Theora into our videoconferencing program. Since a videoconferencing program is realtime, it is UDP-based and wraps the Theora stream in RTP (More on that later). The problem here is that most examples I could find, wraps the theora stream in ogg, and used over tcp or files. I send over UDP and clients need to be able to start in the middle of the stream.

Hi all, something which came up recently in relation to the design of Kate's granulepos was whether or not the granulepos of successive Ogg pages is allowed to be the same, ie. whether or not granulepos must be strictly increasing. As this question is more generally about Ogg granulepos, how about we answer it first and then get back to the discussion of Kate's granulepos ... Here is

Martin Leese wrote: > Subject: [opus] MIME Types and File Extensions > To: opus at xiph.org > Message-ID: > <CAAzqGd_uzR646Nsdt=O2HDxLOYE2=K=5n9UOHLr3Y4BGzdVasw at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi All, > > On the Xiph Wiki page at: > https://wiki.xiph.org/MIME_Types_and_File_Extensions ... > Could somebody more