similar to: Decoding multiple frames at once

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

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

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

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

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

To make it complete, here is the code that I am using to encode a large file: int __stdcall SpxEncode(unsigned char* inBuf, unsigned char* outBuf, unsigned int inlen) { //char *testFile; //FILE *ftest; //testFile = "test"; //ftest = fopen(testFile, "wb"); //fwrite(inBuf,1,inlen,ftest); //take every 320 bytes //copy every short to float

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 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);

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

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

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

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

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

Why does opus_encode return 359, and not 360, please? This is my code: int iShortsRead = fread(m_RawBytes, sizeof(short), (5760/sizeof(short)), fin); for(int i=0;i<iShortsRead;i++) { opus_int32 s; s=m_RawBytes[2*i+1]<<8|m_RawBytes[2*i]; s=((s&0xFFFF)^0x8000)-0x8000; m_ShortsInput[i]=s; }

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 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

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 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

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: >