similar to: Crash bug in CELT plc

We were trying to normalize bands that didn't actually exist (e.g., the last band with 320-sample frames at 32kHz). Thanks to John Ridges for the report. --- libcelt/celt.c | 24 +++++++++++++++++------- 1 files changed, 17 insertions(+), 7 deletions(-) diff --git a/libcelt/celt.c b/libcelt/celt.c index 31d35f8..287c720 100644 --- a/libcelt/celt.c +++ b/libcelt/celt.c @@ -1137,6 +1137,7

Hi all I have spent the last three days evaluating CELT on our supported platforms. I found a bug in quant_bands.c, that due to processor/compilation differences did not cause an issue on x86 platforms, but is a problem on the MIPS processor embedded devices. When decoding on the MIPS devices, there was a lot of noise added during the decoding, the noise is mainly in the 15 khz to 21 khz range.

Should call celt_inner_prod(). --- celt/bands.c | 7 ++++--- celt/bands.h | 2 +- celt/celt_encoder.c | 6 +++--- celt/pitch.c | 2 +- src/opus_multistream_encoder.c | 2 +- 5 files changed, 10 insertions(+), 9 deletions(-) diff --git a/celt/bands.c b/celt/bands.c index bbe8a4c..1ab24aa 100644 --- a/celt/bands.c +++ b/celt/bands.c

--- libcelt/Makefile.am | 6 +- libcelt/bands.c | 26 +++++++++- libcelt/bands.h | 2 +- libcelt/celt.c | 23 +++----- libcelt/pgain_table.h | 133 ------------------------------------------------- libcelt/quant_pitch.c | 117 ------------------------------------------- libcelt/quant_pitch.h | 44 ---------------- 7 files changed, 37 insertions(+), 314

--- libcelt/bands.c | 43 ++++++++++++++++++++++++------------------- libcelt/bands.h | 6 ++---- libcelt/celt.c | 6 ++---- 3 files changed, 28 insertions(+), 27 deletions(-) diff --git a/libcelt/bands.c b/libcelt/bands.c index d0c3498..315cce8 100644 --- a/libcelt/bands.c +++ b/libcelt/bands.c @@ -247,23 +247,6 @@ void compute_pitch_gain(const CELTMode *m, const celt_norm_t *X, const

Has the requirement for FrameSize decreased? I cannot seem to instantiate a mode with a frame size smaller than 120. I have been using a framesize of 100 on previous versions. Thanks bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20110408/af13a1c6/attachment-0002.htm

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

Hi All, Is the following loop in function compute_ebands meant to work out the nearest celt frequency bin to a bark critical band? ?If so why an offset is required to be considered? Thanks in advance if somebody can explain it. Thanks, Hamid /* Spacing follows critical bands */ for (i=0;i<high;i++) { int target = bark_freq[lin+i]; eBands[i+low] =

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

Hi guys, I'm currently working on a CELT implementation for FMOD, and one of the things we want to do is have all memory required allocated beforehand. This is before we have any information about what sounds will be played. We would like to create all of our CELTDecode instances up front. Because celt_mode_create() requires a sample rate and number of channels passed in, this

Hi, I've been trying to encode a live audio input from the microphone on iOS device using opus. Uncompressed audio recording works fine with http://theamazingaudioengine.com/ Then, when I tried to do encoding, I'm stuck at figuring out why the buffer is empty: static int ec_write_byte(ec_enc *_this,unsigned _value){ if(_this->offs+_this->end_offs>=_this->storage)return

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);

I didn't expect this, is it normal? When you use inbandfec, and have packet losses, you end up with more audio samples than you started with. With short frame (2.5), the FEC isn't supposed to do anything but it shouldn't do that should it? and 20mS frame, should do 'something' but? add audio? I'm using opus_demo for this. Here's the steps (I have my test audio in

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear

I have been using the Speex NB mode with an 11025 kHz signal, without adjusting the framesize, and it's been sounding just fine to me. However I was wondering what the best way to make 11025kHz would be. I also have an audio driver that uses 2kB buffers, and the NB framesize of 160 doesn't divide evenly into that so I end up having to shovel tails of buffers around which isn't too

Hi Jean-Marc, For the residue in the echo_cancel function I pass in a pointer to an array of floats. First I used an array the size of the framesize, but I discovered this causes crashes because the echo_cancel funtion writes framesize+1 floats to the pointer. I find this plus one strange, since almost all things are exactly the framesize. Is this an error in the echo_cancel function or simply

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

Hi I have just written a small c programm which reads in speex files and outputs them as asci text numbers. This is necessary as for writing a hw testbench for decoding tests, since thy only read ascii files :-(.I can't read the bitrate properly from the header file. It's allways -1. Please see example below and attached c programm. I wan't to check for a fixed bitrate (8000).

Hi all. I'm trying to decode speex using version 1.1.10's libspeex with fixed_point enabled. copied the sample in the manual (1.1.11) with minor revisions. While running the program, encountered this warning: "Packet is larger than allocated buffer : 38" when calling speex_bits_read_from (&bits, cbits, nBytes) then my program terminated unexpectedly with errors (Unhandled