similar to: Help recording from PCM stream and silence.

1.) after initializing a ogg_stream_state may I just keep calling ogg_stream_packetin with valid ogg_packet's until no more ogg_packets needs to be added? Or do I need to call ogg_stream_pageout after every ogg_stream_packetin? Iaw may I delay calling ogg_stream_pageout until there are no more packets to be added? 2.) I assume I have call ogg_stream_pageout until no more pages can be

Hi, While building an ogg-vorbis stream encoder, I encountered some problems with silence in the audiostream. The bitrate drops to almost zero, and pages going out less then ones a minute what makes the stream to stop / buffer. In earlier postings I read that I shout use ogg_stream_flush as an alternative to ogg_stream_pageout, in case of silence. My question is this; Is there any

Hi Developers, I have a 16-bit PCM data buffer, I want to write that to ogg file. Could you help me to understand how to write pcm data to the ogg? Thanks in advance. Arun balaji -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20150518/6c8c9a63/attachment.htm

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Erik de Castro Lopo wrote: > > > Josh et al, > > > > I've been tracking down a problem with generating OggFlac files. > > <snip> > > > I have looked into this and suspect that the > FLAC__stream_encoder_finish() > > function needs to finish off the stream using a call to

Hi there, for a streaming application developed for a project at the Society for Arts and Technology Montreal, Canada, we would need to know the exact delay of our ogg/vorbis streams. It seems there are several factors that make up the total delay I get. Is there a way to find out how large the delay introduced by the encoder and decoder is? Or in other words, how many unprocessed samples

Josh et al, I've been tracking down a problem with generating OggFlac files. While investigatint this issue I hacked one of the test files from the test_libFLAC directory of the flac-1.1.3-beta2. The result of this hacking is this standalone C file: http://www.mega-nerd.com/tmp/erikd_test.c Directions for compiling this is in the comments at the top of the file. When the program is

Hi, I have just integrated libshout2 and libvorbisenc into mserv, my music server (www.mserv.org). It all works well. Thanks for a great product. The problem I have is that when there is a pause between songs of around 10 seconds, Winamp (2.91) decides (due to the 64kbps dropping to a 2kbps stream I guess), that it needs to re-buffer. It fails miserably and generally dumps the connection, or

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.

I was wondering. When doing ogg only encoding, how do you mark end of stream? I am setting ogg_packet.e_o_s and then submitting it via ogg_stream_packetin() and then, because its the last potential page ogg_stream_flush()ing it. But when I walk through the file, EOS is not set on that logical bitstream. I have been using oggenc.c/encoder.c and oggdec.c/decoder.c for examples, but because the

hi, in my quest to grok ogg i tried to write a small program which extracts ogg_packets, and stuffs them back into ogg pages (which are then written to a file). extracting the packets works fine, but i have problems stuffing them into a new ogg stream. pseudo code: ogg_stream_init(os, serialno) foreach p in packets: ogg_stream_packetin(os, p) // try to extract a page

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

Hi, I have recently implemented encoding/decoding to/from OggVorbis in my app, with the code being based upon the Libvorbisenc sample code supplied with the SDK. It all works very well indeed (encoding from a PCM file, then playback), except that no matter the length of the source PCM file, the last (approx) 4 seconds of audio is always missing from the Ogg file. I am trying to find some

Hi list, I have a very anoying problem when streaming ogg vorbis: due to the variable bitrate encoding data rates sometimes drop to a very low value when there is no audio input signal. After a while the server (Icecast2) just drops my stream because no new ogg pages arrive. For streaming I use a software that I have written but I can't find a way to tell it to never drop data rate below a

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 everyone, I'm trying to build theora encoding into my game engine. Essentially, I'd like to do something similar to png2theora.c, where I take individual screen shots of my game and encode them into a video. Because png2theora.c uses the legacy C API, I used encoder_example.c as a template instead. encoder_example.c seems to be the only code out there that actually uses the th_* API.

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 Mark, Thanks. Let me try the proposed configuration first to make sure that linear 8-bit PCM is retrieved. Regards Amit On Sat, Jan 9, 2016 at 12:28 PM, Mark Harris <mark.hsj at gmail.com> wrote: > 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

Hello All, Turned out to be a coding error. The modified buffer was not being allocated to write back to the File System. After correction the mono 8 and 16 bit for 8K-48K works well now. Regards Amit On Mon, Jan 11, 2016 at 11:05 AM, Amit Ashara <ashara.amit at gmail.com> wrote: > Hello Mark, > > The resulting 8 bit file has a lot of squelching noise compared to the 16 > bit

Hello Benjamin, The original WAV file I have is linear 8-bit PCM. I want to ensure that original linear formats are kept as is. Later I will add support for ulaw. Regards Amit On Fri, Jan 8, 2016 at 5:34 PM, Benjamin Schwartz < benjamin.m.schwartz at gmail.com> wrote: > Do you really need linear 8-bit PCM or do you want ulaw? Linear 8-bit is > ... pretty rare. > > On Thu,

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