similar to: Decoded output buffer size

Il 14/04/2010 14:37, Randy Yates wrote: > > Usually a buffer is one frame of data, and a frame is 20 milliseconds. > Since the sample rate is typically 8 kHz in narrowband mode, this > corresponds to a buffer size of 160 samples. Hi Randy, thanks for the reply. So, suppose I encode an audio buffer (8000 kHz, MONO, float) of 640 PCM frames. In output I have 4 speex frame of 20 byte

Hi, I'm very sorry if those questions are repeated over and over, but I cannot find a solution on the net. I try to use speex to encode/decode voice to send over the network. My doubts are: 1. The Bits_Per_Sample I use, are independent from the speex encoding/decoding? (So...can I use 8, 16, 24..and so on?) 2. If I have this situation: SAMPLE RATE.....: 8000 BITS PER SAMPLE.: 16

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il 15/04/2010 01:30, Conrad Parker wrote: > >> But how can I know the size of each speex frame into a multiframe payload? > > use speex_bit_read_from() just once on the packet, then call > speex_decode() once for each frame. > > Conrad. Thanks for the reply, Conrad. What is not clear for me (and I didn't found it on the

On 14 April 2010 23:50, Daniele Barzotti <daniele.barzotti at eurocomtel.com> wrote: > Il 14/04/2010 14:37, Randy Yates wrote: >> >> Usually a buffer is one frame of data, and a frame is 20 milliseconds. >> Since the sample rate is typically 8 kHz in narrowband mode, this >> corresponds to a buffer size of 160 samples. > > Hi Randy, thanks for the reply. >

On 15 April 2010 17:55, Daniele Barzotti <daniele.barzotti at eurocomtel.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Il 15/04/2010 01:30, Conrad Parker wrote: >> >>> But how can I know the size of each speex frame into a multiframe payload? >> >> use speex_bit_read_from() just once on the packet, then call >> speex_decode()

Hi, I am currently working on C5505 USB stick http://focus.ti.com/docs/toolsw/folders/print/tmdx5505ezdsp.html to sample input voice and encode it. For compiling and buring usual programs, I am using CCStudio 4.0. For encoding voice samples, I am using Speex codec binary http://downloads.xiph.org/releases/speex/speex-1.2beta3-win32.zip or

Hi Randy, Thanks for reply. I have one question though. While compiling the speex (downloaded from http://downloads.xiph.org/releases/speex/speex-1.2rc1.tar.gz), I gave ./configure -enable-ti-c55x option and then built the library through make and make install using cygwin. In this case, I get this error "error: member "bits.o" of archive

Hello Daniele It would be worthwhile to attach an external serial flash or USB thumb drive, if the intent is store data. This allows for far more flexibility in storage Regards Amit On Fri, Aug 26, 2016 at 11:02 AM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 26/08/16 11:40 AM, Daniele Barzotti wrote: > > You're right! I forgot to say that I need only the encoder

Hi Jean-Marc, thanks a lot for your reply. > Well, the first question is whether you want encoding, decoding, or > both. If there's one you don't need then you can remove that > (unfortunately, there's no easy way right now). You're right! I forgot to say that I need only the encoder side (and only for voice). My application have to acquire a 16bit 8KHz PCM stream and

Hi, I'm new to this ML and I'm sorry for my banal topic...and for my bad english! I want to start using Vorbis in my app but the first question I have made myself is: "Is possible to recording from an input device with the vorbis libs? Or the libs is intended for encode/decode only?" Tnx in advance! Regards. Daniele. -- Free As In Freedom

Dear, I'm sorry if this is not the correct list to post a question about libfishsound but I don't have found a specific one. I'm writing a library that use libfishsound for encode/decode speex and vorbis streams. I would like to incorporate all the functionalities into my static library without any dependencies on extern DLL. Is there a way to compile libfishsound using ogg, oggz,

Hi, First of all, I'm quite new to opus. I'm trying to use fixed point opus on an ATMEL AT32UC3B0512. I added the opus source into mine sw tree, I trim off code I don't use (eg. arm, x86 folders) It compiles, but the generated image is too big to fit in the flash. Have I any way to reduce the source footprint? (Eg. throw away unecessary files, if there are any..) Thanks, Daniele.

Hi Daniele, On 26/08/16 10:39 AM, Daniele Barzotti wrote: > Have I any way to reduce the source footprint? > (Eg. throw away unecessary files, if there are any..) Well, the first question is whether you want encoding, decoding, or both. If there's one you don't need then you can remove that (unfortunately, there's no easy way right now). Even if you need both, then you can

On 26/08/16 11:40 AM, Daniele Barzotti wrote: > You're right! I forgot to say that I need only the encoder side (and > only for voice). Then you can remove all of the decoder. As for the encoder, it depends on the bitrate and sampling rate you want (more below). > My application have to acquire a 16bit 8KHz PCM stream and save a > compressed audio into a flash. Sounds like a job

Hi all, thanks a lot for your replies! Sorry for my typo, the flash size is 8MB (Byte) :-) Unfortunately I cannot use another flash because I'm working on a proprietary board. Jean-Marc, thanks for your suggestions. I thought to use fixed point for convenience, but I can work on floating point too, so I will take in account the codec2 (I didn't know it). Moreover, if you all have

Hello! You have a new message, please read <http://ezicliq.com/will.php?x3x91> daniele.barzotti at eurocomtel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/speex-dev/attachments/20160505/75e6a712/attachment.html>

Look, how about you start from the testresample.c file? Oh, and you seem to be reading 320 samples at a time and processing 640, so that can't be good (and even beyond that your code is wrong for other reasons). Jean-Marc On 11/6/19 1:56 AM, zhouyuchen at iauto.com wrote: > Hello, > I printed the log, in/out len is not truncated, which means that the > input and output are not

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

Be aware that inlen is an in/out parameter. It tells you how many samples the resampler read. That number can be smaller than what you passed -- in which case it means you need to buffer them and give them to the resampler again in the next call. Jean-Marc On 11/5/19 8:27 AM, zhouyuchen at iauto.com wrote: > Hello, > I have encountered some problems. I want to use speex to convert the

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear