similar to: About latency, PCM and clients compliance

Hi, does anybody know some info about posting metadata to ices2 via stdin? My system is based on a PHP-script wich sends PCM-data from the player (mpg123 or ogg123) over a pipe to ices2. As i understand the documentation, metadata is sent as normal clear text within the PCM-data when using stdin. I tried the following format: \n artist=foo\n title=bar\n (\n is, of course, a normal

opus_decode() produces 16-bit signed linear PCM, and opus_decode_float() produces 32-bit floating point PCM that is useful when you want a higher bit depth. If you need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed integers for 8-bit linear PCM. (It uses signed for

Hello Mark, The resulting 8 bit file has a lot of squelching noise compared to the 16 bit output from OPUS decoder. During encode I am using popi16fmtBuffer[ui32Loop] = (opus_int16)pcRdBuf[ui32Loop]; And during decode since the data is in the lower 8 bit I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara

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'm running Ices 2.0.1 in conjunction with Icecast 2.3.1 (on gentoo linux) to generate a live ogg stream for radio station WTJU (http://wtju.net/). In addition to encoding and streaming live audio, we'd like to record our broadcasts to disk for archival. I know that ices has a "savefile" feature, but that isn't so useful, because the file being written simply grows

On Sunday, 24 June 2001 at 22:24, Jack Moffitt wrote: > > I decided to make a few shell scripts that can be used to connect to > > icecast, decode/reencode a stream, and then send the result back to icecast. > > > > I guess this can be done with liveice, but this seemed like a simpler > > solution for my needs. I have it triggered by a cgi script that i click

Hello, Yesterday evening, I installed icecast kh13 with your latest libshout. We have to wait 1, 2 or three days to see what is happening... Yes, the samplerate 19404 is exactly what I asked ices to do... Just a compromise between bitrate and samplerate to have a better sound quality. Regarding to that, I didn't found the option to set the ouput rate in ices kh47, and also I was wondering how

Hi, I am using Speex to compress speech samples used for dictation. I am thrilled by the compression ratios Speex achieves with almost no loss in quality. However, I have come across a few files which Speex (I use the Windows variant) refuses to compress. I have put a sample file at http://leggewie.biz/test.wav. The error message I receive is "Unsupported WAVE file fmt chunk, not

When I first install wine using the default repository in Ubuntu 9.04 and run winecfg I get the following error: wine: created the configuration directory '/home/bill/.wine' err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element fixme:system:SetProcessDPIAware stub!

Hello, given a Speex-encoded ogg file, is there a way I can compute the total pcm size (that is, the number of samples after decompression)? I know I can get the frame size from the decoder with SPEEX_GET_FRAME_SIZE, and that the number of frames per packet is written inside the header. But this brings me to the question: how many packets are there in the file? I guess this is more related to

Hi, gurus! I'm now using PCM codec in my application and plan to apply AEC to my app. My question is whether I can use AEC module of speex. AEC module of speex can work for PCM codec, too? Thanks, Jinzhe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20111003/db09011e/attachment.htm

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

Hi! I've spent the last few nights digging into the Vorbis source and working to implement a vorbis_synthesis_pcmout_int() function that kicks out interleaved int16_t pcm data. I think its important to have this function available to make the job for people using the codec a little easier. This function abstracts out the conversion to int16_t and removes the extra overhead of moving the pcm

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

Dear all, I have one simple question. I understand that speex_encode and speex_decode takes float * as an arguement to encode and decode the sound. However, when I get the PCM data from the sound card under win32, it is a just 16 bit array. May I know how do I convert this 16 bit value to speex float format and to convert back? Is there got any routine to do this? YueWeng

I am having trouble converting a .flv file uploaded from Adobe Flex / Flash Player 10 to a Red5 server using the speex coder: http://livedocs.adobe.com/flex/3/langref/flash/media/Microphone.html http://jira.red5.org/confluence/display/codecs/Speex+Codec Questions: 1. How do I extract the audio track out of such a .flv file? 2. How do I convert it from Speex to .wav PCM? Thanks.

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hello: I am running the icecast2.3.2 server and ices0.4 with a perl script which provides the filename of the next track to play. I am reencoding in advance using ffmpeg, which has been very reliable. My main stream is mp3, 64kbps / 44.1 kHz (mono) and I have pre-reencoded the tracks to 32kbps / 22.05 kHz (mono), for a lower data rate stream. At this time, I have not figured out how to get a

Hello Ralph, > Likewise opus_encode() takes 16 bit samples, so you need to extend each > sample from an 8 bit source before encoding. Two questions 1. In opusenc.c which API does the extending the 8-bit to 16-bit? 2. If that is the case then how will 24 bit PCM sample work? Regards Amit On Thu, Jan 7, 2016 at 12:21 PM, Ralph Giles <giles at thaumas.net> wrote: > On 07/01/16

* John McHarry (jmcharry@whqr.org) écrivait : > Have you tried aumix from the command line? I have used it in RH 9 and > Fedora, although not with ices. Yes, I did : in fact, aumix from the command line behaves exactly like the KMix GUI : when I change the PCM or Mic level with aumix, KMix shows the new settings. But even with aumix set to the good values (PCM to 100%), ices get no sound.