similar to: input format

Hi there. I am having a little trouble understanding the frame sizes chosen by the codec. testenc_uwb.c from the speex-1.0 source distribution has a framesize of 640 hardcoded and makes use of this value exclusively. However, a mode query on the actual codec returns 320 as a framesize for this mode. int tmp; speex_mode_query(&speex_uwb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);

Hi, I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE as in speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, &m_SampleRate ); I get 2 The encoder is initialized with m_pEncoderState = speex_encoder_init ( &speex_wb_mode ); On the decoder side, the correct sampling rate is returned: m_pDecoderState = speex_decoder_init ( &speex_wb_mode );

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

First of all, Flash Player can only publish Speex at 16 kHz. 20 ms of audio (320 samples) will result in compressed payload size of 106 bytes (42.4 kbps). When Flash Player sends a TC message, there is 11 bytes TC message header and a single byte of audio message header. For more information, please see ActionScript 3 reference http://help.adobe.com/en_US/AS3LCR/Flash_10.0/ Jozsef

It looks like I'll need to go further into the guts of speex to do this. I do, however, see some lines in nb_celp.c/nb_decode() that look interesting. I guess I'll play with it. I doubt that it will be terribly clean, though. Jean-Marc: Take a look at line 1195 in nb_celp.c (CVS). It reads "speex_warning ("More than to wideband layers found: corrupted

Hi, No, it only happens with speex_wb_mode. But like I said, it is because in wideband mode, the function sb_encoder_init is called and this function initializes the sampling rate using speex_*de*coder_ctl instead of speex_*en*coder_ctl In narrowband mode, the sampling rate is just initialized to 8000 st->sampling_rate=8000; BTW, I saw that my email was added to an existing thread instead of

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

Le jeu 20/02/2003 à 12:21, X-Fixer a écrit : > just to check that I've got everything right. > the encoder allows as input > - mono only Well, there's support for simple stereo (intensity-based) encoding too, but it's probably not necessary to take care of it now. > - 8 or 16 bits as floating point numbers (without scaling to 1.0). > floating-point wavs (IEEE) will

Hi, I'm not sure if I should post this question to this list. If not; please tell me. Ok, here it comes: Is the following code correct for compressing audio? The output I get is so extremely small, but what is more important: if I pass it through zlib, it gets at least 50% smaller! Init: int bitrate = max_bandwidth * 8; speex_bits_init(&libspeex_bits); if

Hello speex-dev! First at all I want to tnak you for this great software codec. It makes many things possible. I am a software developer. One of my projects is NetSpeakerphone - network communication programm (text messaging, voice communication, voice conferencing) for win32 platform. This project's age is about 2 years and many things works just great. Very small voice latency and time shift

Hi, I'm trying to use Speex for real time encoding in an iPhone project but I'm having latency problems. I appreciate any tips you may provide for speeding things up. The recording callback is providing 1024 samples (mono, frequency: 44100 Hz, of type short) every 23 ms (1024.0 / 44100.0). I'm saving the samples in a buffer and use them in the encoding thread (set for mode: wide

hello, i switched to use the encoder.processData() and encoder.getProcessedData() of jspeex. however it looks to me like a memory leak ... memory usage is increasing very fast and there is no visible stop ... after about five minutes java.lang.OutOfMemory occurs. I think it must be the jspeex component, because before i added jspeex to my app usage was constant at about 5mb. is it possible

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

Hi All, Almost all my question is in the subject. I have written the following small program and I get an error message when I try to use SPEEX_GET_QUALITY. This functionality seems not to be used in speexenc.c or speexdec.c. I would appreciate any hints. best regards, Alain Aubord ---------------------------test.c--------------------------------------- #include <stdio.h> #include

Hi, Thank you. I will try it. Do you know some G.72x, GSM, and iLBC optimized for Blackfin ? I mean open source. -- Best regards, Miroslav mailto:miro@space-comm.com Wednesday, March 23, 2005, 9:05:11 PM, you wrote: JMV> Hi, JMV> As far as I understand, the last patch (for TI C5x) I merged in SVN also JMV> allows Blackfin to work, but I haven't

Hi, Sorry! Marc Gimpel. I wrote it in a false thread before. It was my first time. :-) I have to write a small programm which should be able to record speeches and in addition synchronize the recorded speeches with measuring data. My idea is to record the speech in the normal way with speex and without ogg. If a sync message was received, i write a custom in-band messages using mode 13

int frameSize; speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frameSize); Is frameSize in bytes? If not, what unit is it in? I need to know so I know how big to make each element in my circular buffer. Also, do I need to call speex_encoder_ctl( enc_state, SPEEX_SET_SAMPLING_RATE, &sampleRate ); Depending on sample rate I record my audio in? If not, is there any benefit to

FYI, the Asterisk software PBX <http://www.asterisk.org/> has now incorporated my recent patches to support dynamic RTP payload types. As a consequence, its SIP implementation now supports Speex, so if you have a Speex-compatible SIP client, you can use it to make calls using Asterisk. Some caveats: - Only narrowband (8 kHz) Speex is currently supported; not wideband. (Unfortunately,

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any