similar to: API changes for Speex 1.2

Hi, Speex is progressing and I've started thinking about the next 1.2 release (don't hold your breath). Though the bit-stream won't change, the API likely will. The API for 1.1.x already differs from 1.0.x because the speex_encode and speex_decode now use shorts instead of floats. Now, since I'm changing the API anyway, I thought I might as well fix things that might be annoying

Hi, I'm trying to cover all Speex configuration options to set quality/mode/bitrate in my application and I was wondering if the following all makes sense: - abr_bitrate - use average bitrate and set it using bitrate parameter - SPEEX_SET_ABR - cbr_bitrate - use constant bitrate and set it using bitrate parameter - SPEEX_SET_BITRATE - cbr_mode - use constant bitrate and

Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to speex_encode()? But the samplecode that comes with 1.1.4 shows me to put short in float array, and pass the float array to speex_encode() <p>-----Original Message----- From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org] On Behalf Of Jean-Marc Valin Sent: Saturday, April 17, 2004 6:25 AM To: speex

Hi, I'm about finished developing a QuickTime component that supports Speex (on MacOS X and Windows).. As it is now the user can set complexity (SPEEX_SET_COMPLEXITY) and quality (SPEEX_SET_QUALITY / SPEEX_SET_VBR_QUALITY) and to wether to use VBR or not. Will these options make it possible to produce all combinations of bitrates/qualities? Or should I also use

Le ven 16/04/2004 à 16:39, Kenji Chan a écrit : > Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to > speex_encode()? First, I'd like to ak why you want to use 1.1.4. If you don't need the new features (fixed-point or preprocessor), you should probably stick to 1.0.3, which has received more testing. If you choose to use 1.1.4, you need to send shorts to

I tried feeding in the 3 encoded frame in ONE BLOCK, and calling speex_decode() 3 times in a roll. Only the 1st frames came out perfectly. For the other 2, I got "corrupt" frame warning. I was supposed to get 38 bytes consumed each frame (narrow-band, VBR off). I tried speex_bits_remaining() to peek on the # of bits consumed, and got variable (clearly wrong)#s returned. But if I

openh323 speexcodec.cxx has: BOOL SpeexCodec::EncodeFrame(BYTE * buffer, unsigned & length) { // convert PCM to float float floatData[SAMPLES_PER_FRAME]; PINDEX i; for (i = 0; i < SAMPLES_PER_FRAME; i++) floatData[i] = sampleBuffer[i]; // encode PCM data in sampleBuffer to buffer speex_bits_reset(bits); speex_encode(coder_state, floatData, bits);

Jean-Marc and others, We are using Speex inside our Video Conferencing application and have some questions about the trade-off of complexity vs bitrate. A little background information is needed: 1. For us the actual bit-rate isn't that big a deal. Speex running at 14kbps or 8 kbps is minor compared to the video traffic running at 128, 256 or 384 kbps. 2. We care a little about

I'm confused about the format of the PCM/WAV data for speex_encode & speex_decode speex_encode(enc_state, input_frame, &bits); In the manual, it says "input_frame is a (float *) pointing to the beginning of a speech frame" (for encode) But I've found that in speexenc.c and testenc.c, short* is used instead of float* So, isn't it signed 16 bit samples(if

Hi, Is there are document where the interaction between all the configuration options of speex is explained? Speex seems to have CBR, VBR and ABR. You can also use SPEEX_SET_QUALITY (SPEEX_SET_VBR_QUALITY) and SPEEX_SET_BITRATE which I suppose can't be set at the same time. Is there a list of possible combinations somewhere? I also have another question related to jitter. To minimize jitter,

Speex 1.0.3 uses floats, but the unstable version 1.1.x uses short. Jean-Marc Le ven 16/04/2004 16:04, Kenji Chan a crit : > Im confused about the format of the PCM/WAV data for speex_encode & > speex_decode > > speex_encode(enc_state, input_frame, &bits); > > In the manual, it says input_frame is a (float *) pointing to the > beginning of a speech

Hi Jean-Marc, randoms, I've been playing with the 1.1.x api, more specifically modifying code to conditionally build against the 1.0.x api or the 1.1.x api. Code written for 1.0.x builds, with a few warnings, against 1.1.x. The major API change is of course that the type of PCM buffers has changed from float to int. This causes a few warnings related to speex_encode() and speex_decode()

I thought it would be more update/better/faster, any thing wrong? In the future, would you use short, and forget about float? If so, I use 1.1.4, I wouldn't need to change my code in the future, right? You said, it's unstable, yes, I got compiling errors with MsVC6 and MsVC.net And I modified it a bit to get it compiled. Should I post the errors here? <p>-----Original Message-----

On Mon, Apr 19, 2004 at 02:46:44AM -0400, Jean-Marc Valin wrote: > OK, I'll try to do *something* about it. It'll be either "support the > 1.0 ABI" or "make sure the link will break". While the former would be > nice, I'm not sure I want to really want to force 1.2 to have the same > ABI as 1.0.x. More on this when I have some time to think about it.

Here's the situation: 3 frames are being transferred back to back over RTP. The total payload is known (thus the TOTAL frame size). We'd like to decode all 3. According to IETF RTP Profile (appendix C, of Jean-Marc's Speex Codec Manual, verison 1.0), "it's only necessary to consider the sequence of bits as output by the Speex encoder, and present the same sequence to

Hello, When I decode a recoded sound encoded by speex, it has too much noise. The noise is "slightly" reduced if I set the quality to 10. How do I get rid of this noise? The code is as follows. For encoding . state = speex_encoder_init ( &speex_nb_mode ); tmp = 7; speex_encoder_ctl ( state, SPEEX_SET_MODE, &tmp); // set quality tmp = 10; speex_encoder_ctl ( state,

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

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:

Ohh... Nice! This is new in 1.0.1, isn't it? It doesn't seem to be included in the reference manual yet, though. Thanks! Tom <p>Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > OK, this is how it works: > > The encoder calls speex_encode any number of times and then calls > speex_bits_insert_terminator before sending the bits. > > The

Ok, I figured it out. :) This seems to work: 1) Call speex_bits_read_from() once, specifying the location in memory of the compressed data, and the total length of that data. 2) Keep calling speex_decode() until speex_bits_remaining() returns 0. Then you don't have to keep track of the # of frames per packet, or the size of each compressed frame. It's done magically by the codec.