similar to: Can I join many ogg_packet together and decode it to PCM?

Given ONLY an ogg packet, can one be able to decode the packet to PCM? And how ogg vorbis know the info of it (bitrate..etc) in order to decode it? <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word

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-----

Hi Vorbis-Dev, I'm investigating various WebM/Vorbis bugs in chromium. AFAIK muxing Vorbis inside of WebM does not have an official specification, so I'm using ffmpeg's implementation to try to answer 2 questions: 1. Under what circumstances is it valid to find WebM Blocks containing Vorbis data with zero duration? (This would mean the next Block in the Cluster has the exact

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

I am wont to decode a speex 11kbps 8kHz 16bit to a raw data 8kHz 16bit LSB on a c5509. Trying to understand the "testenc-TI-C5x.c" exsample, but it looks to me wary complicated. Is there more documentation for the exsample or a decoder exsample available? Can somebody help? Peter -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I am trying to write a media player in c# which will be able to play .ogg video files which contain Theora video and Vorbis audio streams. Currently I am using LibOgg and LibTheora (via a SWIG generated wrapper). This currently is able to decode the video and output it to the screen. I would like to use LibVorbis to decode the audio stream. Looking at the documentation it looks like I have to use

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

> > Which part? > > Well, my biggest problem is dealing with files. As you have mentioned > that fill_buffer() is obsolete, what has replaced it? ogg_sync_buffer() > didn't seem to be what I was looking for, as far as I can tell... am I > headed in completely the wrong direction? Ah, OK, I understand where you're headed now. The libvorbis API is different than

Once upon a time Monty said: >id3v2 will not be a part of Vorbis. Ogg bistreams allow mixing streams of any >type, and there will be an XML stream type defined for metadata. This is a >better solution than id3 in just about every technical sense. The Ogg >bitstream code to support this already exists. Can you please elaborate? Is this functionality complete? The only mention to

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

> 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? YES, kindly post the errors that you encountered. - Abhishek <p><p><p>---------- Original Message ----------- From: "Kenji Chan" <adslbqmr@tpg.com.au> To: <speex-dev@xiph.org> Sent:

Hi, Searched in Google, browsed xiph vorbis sites but found no good information about how to get the current playtime out of the granuleposition of the ogg_packet. Anyone knows if there is any 'Programming with libvorbis' documentation out there, cause on xiph's site (http://www.xiph.org/vorbis/doc/) there's nothing :( Any help? THX, Tom

I'm trying to encode a picture from my webcam using the theora codec. The final "product" must encode a frame, send it over the network and decode it on "the other side". For now, it must only work locally so we don't care about the transmission. This is what I've understood till now: I have a char * buffer from the camera RG24,I convert it to YCbCr 4:4:4, I

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

Hi. I'm trying to implement Ogg/Vorbis support for the MPEG4IP project. The goal is to support Ogg/Vorbis audio for MPEG-4 streaming. So far I have managed to make the encoder save Ogg packets as an Audio object in an .mp4 file. As a side effect, it can also save Ogg pages in an .ogg file playable by xmms, but that's no big deal. So what I'm doing is this. First the init part. a)

Hi. I assume that I am deeply stupid and have missed something obvious, but why does this bare-bones code not generate a valid (If trivial) opus file? Running opusinfo I ge the following which I interpret this to mean that not even the first packet gets a pass: New logical stream (#1, serial: 1f0cce42): type opus WARNING: Could not decode Opus header packet 0 - invalid Opus stream (1) WARNING:

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

So, I abandoned the hope of using the ogg python bindings to do pure ogg container encoding. I started looking at the libogg in the hopes of retooling the bindings to follow a better object model and it actually looks like the problem is down in libogg, not the bindings. Am I crazy or does libogg rely on libvorbis to return ogg_packets, and that there are no functions that will build an ogg_packet

I'm now trying to use an m3u formatted list to pass to ezstream rather than stdin. Since all of my audio files are in .wav format, and I have oggenc, I see no reason to decode them before ogg encoding them. Is there a configuration I can use to make this happen? I tried omitting the decode parameter and ezstream complained. Here is what I have now: <ezstream>