Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7535 discussions

by hermann meyer

Hi I would save a internal audio buffer to file on exit, to reuse it after e new start. Currently I use stdio fopen/fwrite/fread and save the binary data from the array. That works well. Now I've started to play with libsndfile, first I use SF_FORMAT_WAV | SF_FORMAT_FLOAT which work as nice as stdio. The file size is the same then with plain binary data. To save some bytes on the disk I tried SF_FORMAT_FLAC | SF_FORMAT_PCM_24 which reduce the size nearly to the half. The drawback is that the floats in the internal buffer could go out of the range from -1.0 <-> 1.0 which leads to crackles when the buffer get refiled. As long the values are in range, flac works very well. I tried it with the sf_command (sndfile, SFC_SET_NORM_FLOAT, NULL, SF_TRUE) for write and read, but that didn't help. Yes, I use sf_write_float to write/read the flac file, even if flac didn't support floats. When I understand the libsndfile api right, libsndfile will handle that. So my quetion will be, which is the common format to write flac file with libsndfile, and is there a way to write values out of range into a flac file? regards hermann

12 years, 3 months

[LAD] Fwd: Measuring phase and frequency response of a filter.

by Rafael Vega

Re-posting. Maybe I can get some responses here :) ---------- Forwarded message ---------- From: Rafael Vega <email.rafa(a)gmail.com> Date: Sun, Nov 3, 2013 at 3:31 PM Subject: Measuring phase and frequency response of a filter. To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org> Hi. I'm building a little ear training application for eqing for which I'm building some filter banks in puredata. Can someone point me to a practical way of measuring and plotting my filter's frequency and phase responses? Jack apps, pd patches or code should be fine. Thanks! -- Rafael Vega email.rafa(a)gmail.com -- Rafael Vega email.rafa(a)gmail.com

12 years, 3 months

[LAD] limiting email traffic

by Robin Gareus

Hi all, Due to excessive posts - in particular on the linux-audio-user email list, we are going to rate-limit the number of emails to 13 emails per user per 24 hours with direct follows-up replies to self counting twice. It is not unreasonable to assume that a single user posting more than 12 emails per day is either involved in a heated flame-war or otherwise disrupting reasonable discussion. Neither of which is acceptable on the LA email lists. We understand that some communication requires rapid exchange of many small snippets and recommend users who need to regularly post short messages many times per day to rather use services such as twitter, IRC or simply private email. The LA email lists are mainly intended for discussion, questions & answers, sharing and preservation of knowledge. Note that there are currently over 3000 persons subscribing to the LA lists, every single email sent to the list will result in 3000+ forwards and be archived in various places on the web. Offenders will be notified with a warning message with the 10th email within a day and after the 13th email temporarily banned for 6 hours from further posting to the given email list. Feel free to comment on this email here on this list, but official inquires should be addressed to members of the consortium. See http://linuxaudio.org/contact The system will be activated on 31/Oct/2013 if not vetoed. The actual rate-limit may also be adjusted over time to reflect list-behaviour. yours truly, robin

12 years, 3 months

[LAD] LV2: Communicate from the DSP to the UI

by Aurélien Leblond

Hi everyone, In my lv2 plugin, I need to communicate information from the DSP to the UI, but I don't want to break the DSP/UI separation principle (no Instance or Data access). On top of that, I'm using LVTK. The type of data I'm trying to move is a Vector of Float (buffer type Sound in the ttl) - but even basic information on how to transfer a Float from the DSP to the UI would be valuable. What I have do so far in the DSP (run method) m_forge->set_buffer((uint8_t*)p(p_notify), sizeof(float)); //p_notify being the atom port m_forge->sequence_head(m_notify_frame, 0); ForgeFrame p_frame; Atom p_msg = m_forge->write_blank(p_frame, 1, urids.atom_Float); m_forge->property_head(urids.atom_value, 0); m_forge->write_float(1604); //test data i'm trying to move m_forge->pop(p_frame); In my UI, I just want to see if I receive this data In the port_event method, at the moment the only thing I'm doing is: std::cout << port << " - " << format << std::endl; The port is defined in the TTL as follows: a lv2:OutputPort, atom:AtomPort ; atom:bufferType atom:Float ; atom:supports <http://lv2plug.in/ns/ext/patch#Message> ; lv2:index 1 ; lv2:symbol "notify" ; lv2:name "Notify" ; At the moment, the code compiles and runs without crashing (yeah!), but nothing happens (boo!). Would you guys have any ideas? Thanks in advance, Aurélien

12 years, 3 months

[LAD] linux-port of OpenMusic

by anders.vinjar＠bek.no

Hi all. Ive been working on a linux-port of IRCAMs OpenMusic lately, and think its approaching a useful state now. I would very much appreciate it if some of the brave ones out there did some testing and sent me feedback: http://forge.ircam.fr/p/OM6/downloads/label/421/ The tarball you'll find there includes the OM-6.7 image, source code, a README and the various resources it uses (.so's, libs, fonts, icons, help-files, tutorials, other). This beta is not prepared to be installed anywhere, only to be run from inside the directory where you unpack it (type ./OM_6.7_beta_4 ... ). Plan is to make an installable version once the dust settles some. Theres one thing which needs installing to work however, the music-fonts used in the various editors and factories etc. Theres an .rpm here: http://forge.ircam.fr/p/OM6/downloads/get/omfonts-1.0-1.noarch.rpm.zip - which hopefully takes care of the fonts (theyre also available for manual install in the tarball). The .rpm installs them (4 .ttf-files) inside its own folder /usr/share/fonts/omfonts/* The present version uses JACK for audio + midi I/O, and will try to start a JACK server if its not already running. To get any sound from the usual midi-ports, the linux-port uses libfluidsynth, and attempts to load one of the standard soundfonts into it. However you should be able to route midi-output to any client you prefer (with jack-midi support). Thanks for any feedback. Cheers, -anders

12 years, 3 months

[LAD] list all ALSA-devices (including virtual ones)

by IOhannes m zmoelnig

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 hi all, what's the proper way to list all alsa-devices, including those (plugin, slave,...) defined in asoundrc? e.g. on my desktop i have $ cat ~/.asoundrc pcm.jack { type jack playback_ports { 0 1 } capture_ports { 0 1 } } $ cat /proc/asound/cards 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xf4ff8000 irq 42 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xf3fff000, irq 17 but aplay will show me: <snip> $ LANG=C aplay -L -l null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=MID HDA Intel MID, VT1828S Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=MID,DEV=0 HDA Intel MID, VT1828S Digital HDMI Audio Output **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: VT1828S Analog [VT1828S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 2: VT1828S Alt Analog [VT1828S Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 3: VT1828S Digital [VT1828S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 </snip> there is no mention of my 'jack' pcm, *and* there are devices that i don't know where they come from: namely 'null' and 'pulse' (pulseaudio is installed but not running, as confirmed by <snip> $ pgrep pulse $ </snip> so i figure that the extra devices come from system-configuration files in /usr/share/alsa/alsa.conf(.d/*.conf) but why does my "jack" PCM not show up? i know that i can extend my asoundrc to: <snip> pcm.jack { type jack playback_ports { 0 1 } capture_ports { 0 1 } hint { description "use jack as alsa-device" } } </snip> but i don't fully understand why i have to manually add the "hint" (of which i haven't found *any* documentation) in order to make the device show up with char**hints=NULL; snd_device_name_hint(-1, "pcm", (void***)&hints); any clues? fgasmdr IOhannes -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.15 (GNU/Linux) Comment: Using GnuPG with Icedove - http://www.enigmail.net/ iQIcBAEBCAAGBQJSbpcIAAoJELZQGcR/ejb4MdUP/jatXtJ+8OMWVCpWjREoVUqD j73W6ghI16MNHKhOI2DalmZEQM5P3XocBCR0zoG6mzbzIV5TwkvH/tRwjNbvvkoS suinC/7XFB0p0MpaMfzsMmU34ZlvynCalyWL1DBuApxxaFbTeIxHKTQ3/fF5ztpG iM0fyXVKBw7Y0PX4iubl8CeUwmwgvoZ3oV4SqwkgOk8r+cPB/lrkGFl6vrOQxUV0 lQeHdlzPWDSU/QPfuT8OUpN6A4HIjrEMQ2PMQNAOhSBE3N4WJnGxz3La6TeHU3Xu ybKTn70pslyO+bDluwvqygpaMoBRnSfdXW7+JxLfq/0j1ND2FVIRl0RL2aZADPp6 kcq92L+9py5sF48lEVIEFvbTLXEL7uUH1fxPBkbqw2Rde8CY+aPQqDgQvbnDZMTu oN7JEjySB9meJw7RBnAdMQokC6meJDdWJ/pxZuBapvx6+T0m3PaiUZbO5scPSFVt w3j/fA3NipZwZVENJhuiyWO+H1DUbq8lXHYPSSP5YjgyIkzDu36KQigRfSCuZdrb mq/aDXVmC0+j5EWF9lkDYSDwPFj5GqcsuehX86mAS1LLTyL8/IeyjYTVwbN8FjuJ Sb8g+PNFWJDIAo9OrgOEvzYkQ+qbCAf5TBVWKCWe5HdCBlChpXd8yDMKb52dtB7L GCF4kejxZn75aooUxru5 =1pxG -----END PGP SIGNATURE-----

12 years, 3 months

[LAD] raspberry pi audio in

by Robert Jonsson

Hi guys, As those interested know the raspberry pi lacks an audio input. Now, I came across this some days ago: http://www.noiseisgood.co.nz/?p=365 Apparently it is possible to wire an i2c adc/dac to the raspberry pi and utilize some already available alsa driver. Just thought I would share and ask if anyone has tried anything similar? Regards, Robert

12 years, 3 months

[LAD] Looking for library to categorise audio

by Philipp Überbacher

Hi there, I'm looking for a library that I could use to categorise audio. Relatively simple categories would be sufficient, for example silence, music, speech, noise. It would be preferable if I could use it from a general purpose language over a specialised one (rather java or C than pd). Do you know or even have experience with such a library? Thanks in advance, Philipp

12 years, 4 months

[LAD] Gain in dB

by André Garnier Coutinho

Hey guys, Which definition do you use when you want to do a gain control in dB? GdB = 20log(Out/In) ?

12 years, 4 months

[LAD] LV2 Oscilloscope

by Aurélien Leblond

>> problem is get_supplied_options() returns 0 options when called from the UI. >> > get_supplied_options() should work fine both in UIs and Plugins. My guess > is the host only gives the sampleRate option to the Plugin and not the UI. > Which host are you working with? I'm using Ingen - I sent a ticket to David Robillard onto http://dev.drobilla.net/timeline and will wait for his advise. In the meant time I'll try and debug Ingen itself and see if I can create a patch. > Also, you can compile the UI with a -DLVTK_DEBUG=1, and the console will > print the validation status for any features passed in by the host. > > You could also from the UI send a patch get message and make sure the > Plugin knows how to respond appropriately. Thanks for the ideas - I'll give those a try. Aurélien

12 years, 4 months

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Linux-audio-dev