similar to: A few questions about libvorbis from a newbie

I am deeply sorry about the corrupt message just being sent; there seems to have been a compatibility issue with my mailer and my browser. This is an identical copy of the previous message: Hello Martin, Vorbis encoders are lossy, which is in a sense equivalent to converting the sample size of the raw PCM stream into something that would result in the desired bitrate. The "sample size"

Echoing Hiroka, who is likely on the right track. Simply telling the encoder 'one channel' and then passing stereo data will not convert the stereo data to mono. The encoder encodes whatever it is given... Monty On Tue, Jan 3, 2017 at 6:51 PM, IHARA Hiroka <ihara_h at live.jp> wrote: > Hello Ross, > > The attachment appears at least to me as a valid monaural Vorbis

Hello, I have three questions, though all of which already mentioned by other people in the mailing list, about the accuracy of the Vorbis I Specification published on Feb 27. 1. On page 34 section 4.3.4., explanation of the situation where there are only one submap is omitted, and it is difficult to understand that the decoder should assume in such cases that all the audio channels belong to

Hello Ross, The attachment appears at least to me as a valid monaural Vorbis audio. It says in its header that there is only one audio channel, no stereo mapping is in use, and there is only one audio vector per frame. Actually it does not sound well, in a way that makes me wonder if two channels are unexpectedly merged into one... (thus it might sound like half-speed) I guess there is

Sorry about the personal reply Ian. I hope keeping the subject the same is the way to reply to the list (I have never used a mailing list before). The last thing I am not grasping is how *pcm_total *works without a concept of sample size? If the sample size is whatever I decide to read it as, how does it calculate the number of samples (of one channel I'm assuming because we have to multiply

I’m using a Windows development component which uses vorbis.dll, ogg.dll, vorbisenc.dll for encoding an Ogg Vorbis file. It's all working well except for one user occasionally has a 1 hour file appear as 2 hours and it plays at half speed. It is being converted from stereo to mono before feeding the encoder with a channels=1 configuration. Here is an example file which will be available for

First off, I don't even know if this is the right place to ask these kind of questions, but I haven't been able to find answers anywhere else, so 1. I have found that "pcmTotal * vorbisInfo->channels * 2" gives the uncompressed size of every ogg vorbis file I have used. What is a more robust way of getting the full uncompressed file size? 2. How do I read a certain amount of

I have a wav file which has been converted to an ogg file with libvorbis via ffmpeg. However I've tested a few files just to make sure. I run a standard while loop to using ov_read to pull all the data out of ogg file. Using 4096 as the size to read each call. Watching my debug logs it is reading 2048 bytes each call. When I load the wav file the data chunk for the pcm data is 167680

I was looking at "trunc" to downcast a signed integer, say sint32 to sint16, but it seems to handle unsigned integers only. How do you downcast a signed integer? P.S. This question is for my "Mapping High-Level Constructs to LLVM IR" document :-) -- Mikael -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have this code to decode ogg data: unsigned long PSS_OggStream::DecodeOggVorbis(OggVorbis_File *psOggVorbisFile, char *pDecodeBuffer, unsigned long ulBufferSize, unsigned long ulChannels) { int current_section; long lDecodeSize; unsigned long ulSamples; short *pSamples; unsigned long ulBytesDone = 0; while (true) { #ifdef WIN32 lDecodeSize = ov_read(psOggVorbisFile,

I am trying, for the first time, to decode and play an ogg file with two buffers. My audio is choppy and I can't see a problem on the wavOutxxx end of things. So I am wondering about how I call ov_read. From my TRACE statements, the read seems to work perfectly, eventually returning zero. Is it a simple matter of making the same call to ov_read, aside from changing the buffer and

Hi! I solved the problem using ov_read_float instead ov_read. Still do not know why ov_read returned garbage, but ov_read_float works fine! Thanks! El 17 de julio de 2011 10:41, Juan Miguel Mart?n Mu?oz <jmcubo at gmail.com>escribi?: > Hi, thanks for your replies! > > ogg.k, I'm sure MACOSX is defined. When I'm debugging line by line the > program runs into ov_read.

Hi All, I am trying to program the 'Tremor' decoder onto an array of Processors. I am using four Processors. I did this experiment: I split the while(!eof){} in the main() in to four tasks using if(!eof) statements. In this modification, each if() decodes one packet of data at-a-time, sequentially. Then, I ran the code on a single Processor and the decoding was successful. The

The documentation for ov_read states: "ov_read() will decode at most one vorbis packet per invocation". Can I increase the packet size in my encoder so my decoder makes fewer calls to ov_read? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20131103/d334e7d2/attachment.htm

Ok here is the breakdown. I am trying to impliment streaming into our game using openAL and ogg-vorbis. In the non-streaming function when I call ov_read on the file I should be streaming the buffer gets filled with the correct data. However, when I run ov_read on the same OggVorbis_File when I read the buffer gets filled with 0's. The buffer has not been zero'd and starts filled

Hi, I've ditched the idea of calling the oggvorbis .dlls, and I'm compiling the 1.0beta3 source into my own dll. After opening as "rb" instead of "r", the ov_open() call works. However, ov_read() seems to consistently read less data than I ask for. My buffer is enough to hold two seconds of data, 16 bit 44k mono, that's 176400 bytes. The first call to ov_read()

Hi, I'm new to mailing lists, so don't argue if I do sth wrong... I'm coding on a C++ game with Vorbis- and 5.1-Surround-compatibily. For two channels (stereo), everything's fine, but when I try to read a 6-channel-ogg, ov_read returns -131 (OV_EINVAL). That's strange, because as the specification says, it only returns OV_HOLE or OV_EBADLINK as errors [1]. Have I missed

Has anyone had experience using ov_read in a thread with streaming directsound buffers? i have tried the following, but it just produces a repeating garbage noise. i would appreciate some help. notes: test.ogg in my code was a song that i converted to the vorbis format (Gorillaz - Cling Eastwood) pcmData is defined like this: char pcmData[4096]; the ogg file and the directsound buffer

> On 2016-Jul-11, at 09:05, Ulrich Weigand <ulrich.weigand at de.ibm.com> wrote: > > uweigand accepted this revision. > uweigand added a comment. > This revision is now accepted and ready to land. > > I'll defer to your expertise on that. Patch looks good to me. > > I guess I'm not fully familiar with some of the C++ language details here. Would you

Hi all, Perhaps a stupid question, but I just need pieces of audio from a file. I know I have to use vorbisfile, and the example compiles, but I don't know how to read the buffer. In fact, it's not even documented (or I'm looking at the wrong place). I assumed it was a interleaved buffer containing unsigned shorts, but that didn't work. I also tried the normal decode example,