similar to: 2 questions, frame size and SPEEX_GET_LOOKAHEAD

> >> 2. What does SPEEX_GET_LOOKAHEAD do? How to use it? >> In speexenc.c, there is following code: >> >> ... >> speex_encoder_ctl(st, SPEEX_GET_LOOKAHEAD, &lookahead); >> ... >> nb_encoded = -lookahead; >> >> Can someone explain what this means? > > The lookahead is the number of samples you need to discard at the > start.

Jia Pu a ?crit : > 1. What to do with the last frame that is smaller then frame size? > During encoding, the last frame is often smaller than the required frame > size. In the sample code, proper number of zeros are padded at the end. > So if I don't want those padded zero after decoding, I assume that it is > up to me to keep track of the number of zeros. Is it right? Right.

Speex tells me that the decoder is always 5 ms, but it says that the encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is there an extra frame of delay in the encoder that isn't otherwise accounted for? John Ridges Jean-Marc Valin wrote: > Quoting John Ridges <jridges at masque.com>: > >> I also need to know the precise delays from Speex but I used

JM, I also need to know the precise delays from Speex but I used the SPEEX_GET_LOOKAHEAD control requests to determine them (plus the "speex_resampler_get_output_latency" function from the resampler). The returned values from the Speex lookahead request don't seem to match with the values you gave Alexander. Am I doing this wrong? Thanks, John Ridges speex-dev-request at

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

On Sun, Jun 19, 2005 at 08:07:50PM -0400, Jean-Marc Valin wrote: > Hi Ralph, > > What speexenc does (Speex itself does not know about Ogg) is that it gives > packet N the granulepos "N*frame_size - lookahead". In the case of narrowband, > the first frame would have granulepos "1*160 - 80", so 80. I can't say about > your example because Speex cannot

Ok, let me first explain why 5ms matters, even they are 0's, in my particular application. I am working on a speech synthesis system. The basic idea is concatenating pre-recorded phonemes or words into longer sentences. So any missing or extra samples, even it is as short as 5~10ms, cause very noticeable discontinuities. I want to use speex to compress/decompress those pre-recorded

I tried to compress 160 samples of random garbage (range 0-500.f) at 4kbps and got a crash in nb_encode -> forced_pitch_quant because start was -858993460 (== ol_pitch). -- Daniel, Epic Games Inc. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org'

Quoting John Ridges <jridges at masque.com>: > Speex tells me that the decoder is always 5 ms, but it says that the > encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is > there an extra frame of delay in the encoder that isn't otherwise > accounted for? Oh, delay = frame_size + lookahead If you have a frame size of 20 ms, then there's no choice but

[At the risk of educating you about something you might already know] Natural speech in most human languages gradually changes from one phoneme to the next. Concatenating phonemes together from a fixed, prerecorded, unflexible set would give rise to abrupt changes between them (both in phoneme quality and in pitch), and thus make the resulting speech hard to understand and/or uncomfortable

On Fri, Apr 4, 2008 at 12:19 AM, Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > Jahn, Ray (R.) a ?crit : > > > Dear Speex codec community: > > > > I am working on conversion of voice files. I could not figure out how to use speexdec.exe 1.2.3 in piped mode in order to avoid the creation of the potentially large intermediate *.wav or *.pcm files. Any

Jean-Marc, What exactly does the speex code do to calculate the granulepos? With vorbis it's the 'count of decodable samples including this packet'. So if you had a packet that allowed you to decode 1024 samples since the beginning of playback, the granulepos is 1024, not for example 1023 which would be the *index* of the last sample assuming C-style array indicies starting at

Quoting John Ridges <jridges at masque.com>: > I also need to know the precise delays from Speex but I used the > SPEEX_GET_LOOKAHEAD control requests to determine them (plus the > "speex_resampler_get_output_latency" function from the resampler). The > returned values from the Speex lookahead request don't seem to match > with the values you gave Alexander. Am I

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, I am new to speex and I am trying to use the sample program given in the speex document. I have made some modification to that sample, so that input can be read from a file and the output can be re-directed to the file. I tried to encode an audio file using the sampleenc and decode the same by sampledec.c I am able to do it successfully, but when i try to play the output file from the

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

Dear Jean-Marc and Peter: Thank you both very much for your time and advice. I did not realize that Lame MP3 code has a -r option (without reading its code). I have tried the suggested command lines verbatim with the added -r option, along with other combination of option settings. I could not figure out how to eliminate the distortion in the result, like voice turning either high or low

Hi, I encoded file in a ipaq 1940, pocket pc 2003, processor Samsung s3c241(similar to ARM), 266 Mhz. I would like to know your opinion for the quality output if this is right to entered parameters. File original: http://padawan.ing.puc.cl/chavezorig.wav File encoded: http://padawan.ing.puc.cl/chavezq1.wav The parameters for encoded are: state = speex_encoder_init(&speex_nb_mode); int tmp=0;

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

I've targeted Speex to run under eCos with Thumb interworking using arm-elf-gcc 3.3.3 I'd written a small test case to verifiy the operation: #include <speex/speex.h> void speex_test(void) { SpeexBits bits; unsigned int frame_size; void *enc_state; float *speex_frame; cyg_uint64 t1, t2; int i; speex_frame = (float *)0x00030000; /* Address of Speex data