similar to: Delay in decoder/playback

I'm using SDL on an ARM based board running Linux (Olimex-AT91SAM9261) . I successfully cross-compiled, installed and tested the libraries (SDL, SDL_mixer, libaudio, libvorbis, libvorbisfile etc.) on the system. The sound output is done thru ALSA (SDL_AUDIODRIVER=alsa). And the ogg file decoding thru libvorbis/libvorbisfile libraries. Now I'm trying to play ogg files. First I used the

Hi everyone, I'd like to present an open-source project I made for use in my company's projects: Theora Playback Library website: http://libtheoraplayer.sourceforge.net It's a multi-threaded library that decodes theora videos, decoding frames in advance (precaching), decodes any syncs audio and provides a very fast YUV->RGB conversion. One of the most important features is that

Dear theora developer community, currently I'm working on a simple Theora player for Windows. But the code in the player_example.c seems not to have the performance of other implementations like the Direct Show filters by illuminate. In the example player, all important things are done in one thread: decoding the next vorbis or theora packet(s) and reading from the physical stream (+ split

I have the following problem. Be an encoder application (station) and a decoder application (receiver). If I feed the vorbis encoder with 'n' pcm samples in the station I want to know how many pcm samples will be decoded in the receiver if I feed the decoder with the encoded output. The ogg_page_granulepos is right for the purpose? I read the documentation about the ogg_page_granulepos

Sounds like either your machine is too slow or you have some other buffer issue. A decode thread is usually ideal so that data can be decoded before it is needed by the buffer fill operation. You might be able to get a sasifactory solution by increasing the number of buffers and using them in a round-robin fashion 1-2-3-4-1-2-3-4. Wouldn't do any more than 1/4 sec per buffer. Cheers, Chris

Hi Ralph, thanks for your posting. Yes, the standard example player in the theora distribution could also do it when theora would not need so much time. I removed the frame dropping from the example because it's based on some audio stuff under Linux that isn't available on Windows. Currently I've no frame dropping handling build in because the first goal is a good raw performance of

Hi, Jean-Marc, and thanks for the quick reply. Let me just say I'm a huge fan of speex, and the work you've done. I actually barely understand what I'm reading so far in the source code and documentation, just enough to understand just how cool the algorithms are. LPC10 and MELP allow me to speed up speech with a simple hack on the decoder frame size. Playing fewer samples per

Hi, I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get linphonec to work with Asterisk. I have the echo test working, but when I dial in to this, to voicemail or anything else using Playback() to play a sample, I hear nothing for ages (10-15 secs) and then little sections. With the echo test, I get the tail of the message (...pressing the pound

Hi Bill, Any attempt to alter speed by simple insert or dropping produces poor results. Even if you can get it to sound smooth, the resulting pitch shift is horrible. You really need to use a transform that alters speed smoothly, while maintaining the original pitch of the voice. If you look in my spandsp library you will find a module which does exactly this, using an algorithm called

I was able to easily hack in an option to play back at different speeds. For example, using "speexdec --speed 2.0 file.enc file.wav" plays back encoded file.enc at 2X speed. What I did was divide st->frameSize and st->subFrameSize by the speedup, and added a SPEEX_SET_SPEED decoder control for the nb_celp decoder. This produced speech that was 2X faster than the original.

Hello, did anybody else notice that current cvs outputs green/blue/blocky garbage in mplayer (player_example doesn´t play at all)? Seems the decoder doesn´t like recent encoder changes ;-) (yes - I know CVS is experimental. I just couldn´t resist *g*) At the time being I can´t compile properly on my gentoo machine: user@notebook theora $ ./autogen.sh I am going to run ./configure with no

Hi, Steve. I tried your the time_scale_tests program, and it works well! Especially for low speed changes, it's the best I've heard so far. For high speed increases, there is what sounds like static added to the sound output. I've attached two sound samples of high speed speech, which is a 4X speed up of a popular TTS voice in the blind community (voxin/Eloquence). I've sped

You're asking the wrong question. The question is not "why does it would bad with Speex?", but "why does it sound good with LPC10 and MELP?". And the answer is that both are vocoders. Try dropping frames/subframes with anything else (Vorbis, MP3, G.729, u-law, ...) and it'll sound terrible as well. The only reason it sounds good with vocoders is because the

Hi - I've noticed that I'm getting audio delays when asterisk is playing back a file from disk and new zap channels are being created or destroyed. Audio streams are generally fine (person to person calls do not experience this issue). Sometimes the drops are very short - barely noticeable. Sometimes they are up to 1 second, and whatever file is being played resumes at the

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally,

Version 1.2.1 introduced new rice coding techniques that are used by the reference encoder for 24 bit files. An older version of the decoder will have trouble with frames that use this encoding... Maybe that's where the strange noises come from... Pyt. On 6 f?vr. 2011, at 06:01, Brian Willoughby <brianw at sounds.wa.com> wrote: > Thanks for bringing up this aspect, Nicholas. I

Whoops, left this message in my outbox. I managed to fix the problem. Apparently I was only copying 160 bytes (Frame Size) back into the audio stream when I should have been copying 320 (chars <-> shorts confused me there). Hence why I could hear myself yet it was distorted. Half the wav was missing =) To answer some of the other questions here, for any insight into what I'm doing:

Here's one clue about whatever is causing the low quality speech. Speech sounds terrible at 1.01X faster, and it sounds excellent at normal speed (1.0X). So, the main problem is something that breaks with any change in frame size in the decoder. Any idea what that might be? Thanks, Bill On Tue, Oct 19, 2010 at 5:14 PM, Bill Cox <waywardgeek at gmail.com> wrote: > I was able to

Speaking of portability and people more familiar with SDL: The reason we have two example players for theora is that monty wrote the original example around the OSS audio interface used on several free unix-like operating systems, which immediately excludes win32 and MacOS, even though it used SDL to display the frames. SDL also has audio support, and is quite widely ported. splayer replaces

I'm getting garbled playback with decoded fragments and I'm hoping someone here can point me in the right direction to correcting the problem. Essentially I'm capturing audio from the microphone. I stream it over the net, but for testing purposes with this API I'm just grabbing the whole chunk and encoding / decoding it right away and then updating the sound buffer for