similar to: Average Bitrate Calculation

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dear list, i am trying to write a small program which reads an ogg file and writes it to another ogg file (and changes serial number, granulepos etc on the fly). reading the ogg file is ok (ogg_sync_pageout, ogg_stream_pagein, ogg_stream_packetout). but writing the file doesn't work - the granulepos and page structures don't match with the original file. here's what i am doing.

2008/10/29 Romain Beauxis <toots at rastageeks.org>: > > I am currently implementing theora for our application. > In our model for generating ogg streams, we may want to stop > a stream while not providing a new YUV data buffer for encoding. > > Current API doesn't allow such thing, since the eos flag is set by the > packetout function only when the last_p parameter

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:

I've been trying to piece my way through the encoder_example.c program to better understand how to encode files as ogg/vorbis. I'm stuck on two sections of the code. This is the first /* uninterleave samples */ for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)| (0x00ff&(int)readbuffer[i*4]))/32768.f;

Ok I just finished writing a Vorbis encoder/decoder, and here are some of my comments (I'm sure some of these are well known, but here they are anyway - sorry this is long): 1. vorbis_info: a. Accessing the pre-computed info structs through pointers to static data is messy IMO. I should be able to vorbis_info_init (or similar) an info struct to be exactly the same as info_A, for

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

I am having some trouble calculating the duration of my speex files. I am writing a speex player in Java and I am using the JSpeex port of the codec which works but doesn't give me the audio duration which I need to show progress or implement a scrubber. JSpeex appears to be all but dead but I don't see any other option for Java playback. I have tried the method posted here:

Hi, I am continuing to try to debug my ogg/theora decoder. The issue seems to be that not enough packets are being pulled from the bitstream, here is some sample output from my player with some debugging comments put in: (playback began at first frame, this is from the middle of the clip) get page at 1765027 got packet frame++, skip is 0 frame decoded page processed, kframe=414 : offs=442 get

I've been looking over Benjamin Schwartz's Skeleton A-mod proposal. I've been pretty busy with other projects over the past few months, so haven't had a chance to look at Ogg indexing until now... In general, I think Benjamin's ideas are sound, they're improvements, and I'm open to being convinced to take them in the next index version. We may as well get the index

I'm trying to learn how to use libtheora but I'm having some problems: I believe to have encoded a video, but I can't decode it. I've tried 2 things: to record a .ogg file and to write a libtheora decoder, using the encoded ogg_packets present in memory. The source code is here: http://opensvn.csie.org/ribamar/projects/streaming/cvaenc.c (it has a makefile in

(I've already posted this message month ago, but nobody answered, may be it was not delivered to newsgroup?) Hello, All. I wrote an encoder-decoder based on example from OGG-Vorbis SDK. This encoder can encode a large amount of small WAV-files with equal parameters into one sound archive. Then I decode this sound archive back into large amount of small WAV-files. Theese files are

>> Running ogginfo on the stream data gives: >> stream_integrity=fail >This line in the ogginfo output concerns me. It means that there are actually >damaged Ogg packets in the stream. How are you saving this stream to disk? This particular one was obtained with wget on OpenBSD. I've observed the same problem with files saved from Winamp. >I think there is a problem

On 2010-04-11 10:09, Randy Yates wrote: > If I specify VBR mode via > > speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_VBR,&isTrue); Try: speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_ABR, &desiredRate);

I would very much like a lowpass option because for FM radio broadcasting I don't want to encode frequencies above 15khz. I'm waiting for this option before switching to ogg from mp3(lame). Ross. > -----Original Message----- > From: owner-vorbis@xiph.org [mailto:owner-vorbis@xiph.org]On Behalf Of > Gian-Carlo Pascutto > Sent: Tuesday, 4 September 2001 01:46 > To:

Jan, Here's a hacky patch to add a few commandline options for setting comment header fields in ffmpeg2theora. It's a bit big because I virtualized the global info struct in theorautils.c. In retrospect that probably wasn't necessary, but I think it's cleaner anyway. I didn't test it because I couldn't compile ffmpeg2theora, but modulo bugs it should support

Hi all, I noticed the following Subversion commit today: r6719 | xiphmont | 2004-05-18 16:04:53 +1000 (Tue, 18 May 2004) | 11 lines Updated doc to reflect current proposal... Not as much a proposal at this point actually; this is the way I'm now implementing it. Although we're still in the 'RFC'/'look for horrible lossage' stage, this is close to being

Dear all I'm aware that there have been several discussions about the seeking issue and I'm sorry to bring this up again. To solve the problem with Inter-Frame garbage, a seek to the previous keyframe has to be made. The keyframe number should be extracted from the granulepos of the frame where we want to seek to. I hope I understood the theory - unfortunately a few questions have

Question: is it permissible to put more than one frame packet into a page? At 30 fps, Ogg page overhead is on the order of 8 - 10 kbits/sec. At low datarates, this is going to be unacceptable. If I shove a few frames into a page, how does this affect the granulepos stuff? ___ Dan Miller (++,) Founder, On2 Technologies <p>--- >8 ---- List archives: http://www.xiph.org/archives/