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

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

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

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

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

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

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

Hi, I was hoping to get some clarification on the meaning of algorithmic latency in Speex. For wideband signals, algorithmic delay is specified to be 34 ms (20 ms is the frame size and 14 ms is the lookahead). Does this number (34 ms) account for the double-buffering on the input? Assuming that the computation time and sound wave propagation time are negligible, should I expect compressed

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:

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'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

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, Andras, Thanks for the comments. Yes, I am aware of those issues. I probably should have been more accurate on my usage of terms. Actually in my project, the unit collection is a mixture of diphones and words. However seems to me, these synthesizer specific issue is irrelevant to my question about speex. As you said, i merely use speex as storage methods. All I ask for is to get the