similar to: API suggestions

> It is unusual to require a different sequence of API calls > depending on whether the signal is Mono or Stereo. > This is especially evident in the decoder where you would > register a callback for the SPEEX_INBAND_STEREO message. > The need for explicitly calling the speex_en/decode_stereo() > function in addition to speex_en/decode() is cumbersome. > All these tasks could

Jean-Marc, The following patch makes the above two read functions const correct. Cheers, Erik diff --git a/include/speex/speex_bits.h b/include/speex/speex_bits.h index a26fb4c..234ec53 100644 --- a/include/speex/speex_bits.h +++ b/include/speex/speex_bits.h @@ -77,7 +77,7 @@ void speex_bits_reset(SpeexBits *bits); void speex_bits_rewind(SpeexBits *bits); /** Initializes the bit-stream

> If I understand what you said correctly, there is such a call: > speex_bits_read_whole_bytes, which adds a couple byte to the SpeexBits > struct while removing the ones that have already been read. Oh, how could I not have seen that one? That is exactly what I need. Thanks for the pointer. >> It is unusual to require a different sequence of API calls >> depending on

hi Jean-Marc, I just updated from svn and it looks like there are some errors in the speex.mmp file. the filters.c is duplicated, and the fft/kiss.c files are still missing.. /alfred Jean-Marc Valin wrote: > Applied in svn. Thanks. > > Jean-Marc > > Alfred E. Heggestad a ?crit : >> hi, >> >> this patch should fix the symbian build. >> >> >>

Hi all, i've to encode voice from a voicemodem. I choose speex 1.0.5 for its quality in voice encoding. I've tried to implement an encoder but unsuccesfully. Here's my code: /* ============ SPEEX stream ENCODER ============================================ */ int SPEEX_EncodePCM(struct _IDA_ClientSocket *IDA,char *buffer,unsigned char *PCM,int num_samples) { /* buffer point to the

Hi, I am using speex 1.2beta2 on a narrowband 16-bit, 8khz system that has a severe program space problem and will not fit speex in its normal operation. In an attempt to shrink speex I placed a breakpoint in every function and ran a decode and encode and removed the breakpoints that I hit. in the functions that had a breakpoint that I didn't hit I commented out those functions (as well as

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.

hi, this patch should fix the symbian build. /alfred -------------- next part -------------- A non-text attachment was scrubbed... Name: speex-symbian2.patch Type: text/x-patch Size: 2669 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20061115/5ac4915e/speex-symbian2.bin

for the bits init I am using speex_bits_set_bit_buffer and I don't use the write to or read from because the data is already in the buffer I am reading from and I am writing to the final buffer so I don't need to move arrays around. what part is the vocoder part of the decode? Thanks for your help! -Mike >>> Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> 08/06/07

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

I'm trying to use speex on OMAP's DSP using dsp gateway, and have some questions. The code almost works (within few next days I'll post a some kind tutorial and patches), but it seems to be a little bug in bits.c. The code looks like this void speex_bits_read_from(SpeexBits *bits, char *chars, int len) { ////////////////// bla-bla-bla /////////////////// for (i=0;i<len;i++)

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, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

Okay.. I love flac but just had a rather serious failure that really shakes my confidence. It resulted in the near loss of a master audio recording. Fortunately, I have a backup. Though there may have been other cases where I have lost original material because I have been compressing a lot of originals and deleting them after doing a 'flac -t' on them. Basically, flac failed with a

I'm considering creating an ogg vorbis decoder chip for my senior project in school. At this point, my idea is to have a program (such as XMMS, ogg123, WinAMP) send the encoded ogg bitstream through a USB port to my project, which is then decoded. My project then returns the raw PCM audio, which is then handled by the player program for final audio output. I'm planning on USB as it

I downloaded test vectors from opus-codec.org site. The site indicated that the test files are intended for verifying that the Opus decoders are operating properly. I assume I could also use them to verify my encoder implementation, right? For example, use testvector01.dec as input to my encoder and compare its output bitstream with the corresponding testvector01.bit bitstream file. Any idea

Hi everyone, I've recently done some major internal changes in Speex aimed at reducing RAM (by nearly a factor of 2!) and improving quality of the fixed-point. In doing so, I might have accidently broken a few things. I'd like to hear feedback on the current svn code to make sure I fix any regression before the next release. I'm already aware that --enable-vorbis-psy is broken and

Hi, I am using the speex 1.0.4 library from Windows. I have posted my problem before but didn't get a solution. I am doing an VOIP project in which i am recording sound and streaming it to the peer. I wanted to encode and decode wav files that brought me to this site. I am recording sound in the following format:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM;

Hi, I'm trying to encode a stereo speex file on the blackfin without much luck. This is a standalone application with no OS using the speex library provided in the blackfin multimedia sdk. I believe this library was ported to run using Visual DSP++ by Jean-Marc (may be wrong). I have mono encoding working fine, to add stereo encoding I just added a call to speex_encode_stereo_int()

Hi Marian, I took a brief look at RFC 5574 and as far as I understand, you can simply do it like this: SpeexBits b; speex_encode(state, some320bytePCM1, &b); speex_encode(state, some320bytePCM2, &b); After that, get the encoded data with speex_bits_write and put it in your packet. Mark -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at