similar to: not hearing the starts of words when encoding

Hi, Do you get that problem with speex_nb_mode too (or only speex_wb_mode)? Jean-Marc Le lun 29/03/2004 à 05:35, Chris Flerackers a écrit : > Hi, > > I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE > as in > speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, > &m_SampleRate ); > I get > 2 > The encoder is initialized with

Hi, No, it only happens with speex_wb_mode. But like I said, it is because in wideband mode, the function sb_encoder_init is called and this function initializes the sampling rate using speex_*de*coder_ctl instead of speex_*en*coder_ctl In narrowband mode, the sampling rate is just initialized to 8000 st->sampling_rate=8000; BTW, I saw that my email was added to an existing thread instead of

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

Hi, I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE as in speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, &m_SampleRate ); I get 2 The encoder is initialized with m_pEncoderState = speex_encoder_init ( &speex_wb_mode ); On the decoder side, the correct sampling rate is returned: m_pDecoderState = speex_decoder_init ( &speex_wb_mode );

Thanks for the advice. With complexity=1, bit rates 5950,8000 and 15000 (CBR only), I'm still getting 30-50 seconds encoding time for a 10-second file. To anyone who has made this work in ARMv4, or knows how to, can I get some advice on the settings? FIXED_POINT is already defined in all relevant files. My source code is patterned after sampleenc and I haven't tried using Ogg.. my source

Speex 1.0 - in file sb_celp.c line 218 change speex_decoder_ctl(...) to speex_encoder_ctl(...): void *sb_encoder_init(SpeexMode *m) { . . . --> speex_decoder_ctl(st->st_low, SPEEX_GET_SAMPLING_RATE, &st->sampling_rate); // Replace <-- speex_encoder_ctl(st->st_low, SPEEX_GET_SAMPLING_RATE, &st->sampling_rate); st->sampling_rate*=2; return st; } --

>jesus: > Now I see that it happens when I specify a framesize in > speex_echo_state_init greater than the one used in > speex_preprocess_state_init. > > Next combination works: > codec framesize = 160 > preprocessor framesize = 160 > eco framesize = 160 > > > Next combination DOESNT work (produces heap corruption): > codec framesize = 160 >

On Sat, Mar 20, 2010 at 12:04 AM, Jozsef Vass <jovass at adobe.com> wrote: > Nicer way: > > void* speexState = speex_encoder_init(&speex_wb_mode); > int speexFrameSize, speexRate; > speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); > speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); > SpeexPreprocessState*

What are you trying to accomplish? My code sample was about how Flash Player microphone input in speex. If you want to do decode, please see speexdec.c Jozsef -----Original Message----- From: Max Lapshin [mailto:max.lapshin at gmail.com] Sent: Tuesday, March 30, 2010 8:53 AM To: Jozsef Vass Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] Speex in flash player: how to work with? On Sat, Mar

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

I found a problem with trying to use the speex plugin for xmms on my PowerBook G3 'Pismo' PPC machine, running kernel 2.4.19. It turns out that the kernel sound driver ignores the endian of the data format flag and instead expects the data to be native endian. This is the same problem that was affecting speexdec. I have now fixed this problem, and another compilation issue that I found. I

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

Oops... You've just found a bug. Seems like you're the first one to use that call. Anyway, it's now fixed in CVS (both trunk and 1.0.x branch). Thanks for the bug report. Jean-Marc Le mar 03/06/2003 à 01:16, Christian Buchner a écrit : > Hi there. > > I am having a little trouble understanding the frame sizes chosen > by the codec. > > testenc_uwb.c from

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

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

Hi, I've looked in various FAQs and web pages for this info, but just can't seem to find it. When two tones of two different frequencies sound equally loud, what's the (rough) relationship between their power? This is for percussive sounds in music, but I assume it's roughly the same for all sounds. The ear seems more sensitive at high frequencies. For example, when you

On 6 May 2023 09:56:35 BST, Hao Chen <chenh at yusur.tech> wrote: >In the current code, if the maximum MTU supported by the virtio net hardware is 9000, the default MTU of the virtio net driver will also be set to 9000. When sending packets through "ping -s 5000", if the peer router does not support negotiating a path MTU through ICMP packets, the packets will be discarded.

Hi Jean-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but I hear

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 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);