Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7509 discussions

by Machel Sylvain

Hi, I'm about to work on a LV2 wrapper for the FxEngine Framework and I would like to know if it exits compliant tests to validate LV2 plugin according to the port type, scope, ect ..... ? With the exeption of a few unitary tests, I didn't find anything on the web. Thanks in advance, regards, Sylvain

15 years, 8 months

Re: [LAD] alsaplayer fix

by Emanuel Rumpf

2009/11/2 <fons(a)kokkinizita.net>: > On Mon, Nov 02, 2009 at 10:31:26PM +0100, Emanuel Rumpf wrote: >> Some fixes for alsaplayer 0.99.80 >> >> - compiles with gcc 4.4 now >> - works with new JACK version (1.9.3) (jackd audio server) > > If you did add a call to jack_get_sample_rate(), that > was not necessary (but it won't harm). > I did. There was a strange message about the sample rate being zero. Too low quality for my taste :-)) > When you enable > the sample_rate callback, jack should call it immediately > with the current sample rate. Jack1 does so, Jack2 aka > jackdmp aka 1.9.* doesn't. So the bug is in 1.9.*, not > in alsaplayer. It has been corrected in SVN. > Thanks for mentioning. -- E.R.

15 years, 8 months

alsaplayer fix

by Emanuel Rumpf

Some fixes for alsaplayer 0.99.80 - compiles with gcc 4.4 now - works with new JACK version (1.9.3) (jackd audio server) Replace the old files with those provided in this package. Recompile. -- E.R.

15 years, 8 months

State of Plugin API's

by Gabriel M. Beddingfield

Hi guys, I'm looking to delve into the world of plugins (for both synth and fx), and of the free API's I find LADSPA, DSSI, and LV2. But the picture is a little confusing. It appears that LV2 is "current," that DSSI is deprecated, and that LADSPA would be deprecated if it weren't so widely adopted. However, LV2 is slow in being adopted. Is this developer resistance... or is it just too new? Also, I can't find any applications that will host all 3, nor even LV2+DSSI. Is there a technical reason for this? Thanks in advance, Gabriel

15 years, 8 months

[semi-OT] midi snakes using CAT5?

by Jörn Nettingsmeier

hi everyone! i'm playing with my shiny new BCF2K, and i'm going to use it some distance from my machine, so i'm going to try a midi link instead of USB. what is the maximum length of midi chain that you have used without problems? i read somewhere that no more than 15 meters are recommended, which strikes me as pretty short even if it's unbalanced. my idea is to abuse a cat5 cable as a triple midi loom - do you think that could work? pinout would be as follows: 1 midi 1 signal 2 midi 1 ground 3 midi 2 signal 4 midi 2 ground 5 midi 3 signal 6 midi 6 ground 7 common +5V 8 common +5V best, jörn

15 years, 8 months

remote dbus and ladish. Was: gets confused when used through ssh -X

by Nedko Arnaudov

Patrick Shirkey <pshirkey(a)boosthardware.com> writes: > I don't see much point of storing a local copy of a ssh tunneled config > file. If you are tunnelling the config file should be accessed from the > remote machine. It's different if you are using multiple qjackctls on > the same desktop and connecting to local jackd instance/s as well as > netjack'd instances on remote machines. That seems like a heavy deal and > as Nedko has mentioned previously it can actually be managed by the LADI > tools now although that requires running dbus too which is not to > everyones liking. Yesterday, I've booted my wife`s windows machine from an Ubuntu LiveCD and made some remote dbus tests. There are two options to initiate this through ssh. The first one is to start socat[1] at both sides and use tcp for connecting them: dbus app <-> unix socket <-> socat <-> tcp over ethernet <-> socat <-> unix socket <-> dbus daemon <-> dbus app The other one is to use gabriel [2] at client side. Gabriel uses libssh [3] to setup socat remotely and tunnels the dbus traffic over ssh tunnel: dbus app <-> unix socket <-> gabriel <-> ssh over ethernet <-> socat <-> unix socket <-> dbus daemon <-> dbus app Gabriel`s approach is of course better in long term, because dbus traffic is encrypted and because it sets remote part automatically. Bad news about gabriel is that last commit is from February 2007, it needs a patch to work with latest libssh (0.3.4) and it allows only one client to use the local unix socket. Both approaches have restriction because of the EXTENRAL dbus authentication mechanism. That mechanism sends unix user id and it is matched remotely. So in order remote dbus to work, uids should be same. Of course this is lame and I already have plan for this: the client side (gabriel/ladish), can get the remote uid through ssh and change the uid "token". Plan for the remote capable ladish is to take the gabriel`s approach. Requirements for the remote (aux) hosts will be - ssh server, dbus, jackdbus and socat. Requirements for the local (central) host will be ssh client, libssh, dbus, jackdbus, ladish. ladish will provide dbus service that will act as manager for remote dbus connections. Such service could be reused for other remote dbus purposes and as such could be a separate project (or could be made as part of gabriel). Multihost ladish is planned for the 2.0 release and wont happen anytime soon unless someone with high motivation steps to work on this. I'd like to ask users of multiple-jack-servers-on-same-host (Fons?) to explain what it is used for and how it works. As I personally dont use such setup I can't make ladish suitable for such setups without feedback. Same applies for remote jack management and netjack but I've gathered some knowledge because of the obvious netjack popularity. [1] http://www.dest-unreach.org/socat/ [2] http://gabriel.sourceforge.net/ [3] http://www.libssh.org/ -- Nedko Arnaudov <GnuPG KeyID: DE1716B0>

15 years, 8 months

Audigy soundcard silent after upgrade to Ubuntu Studio

by David Morrell

Problem solved by a contributor on LAU. He sent me a /var/lib/asound.state file from a working Audigy2 installation and, what do you know, it worked. There is a file that describes the state of the sound card in /var/lib/alsa/asound.state. When the original (bad) version was replaced with one from a working Audigy2 sound cared (good version) the card started working. The replacement asound.state file has 227 controls vs 216 for the oroginal. The following 14 switch controls existed only in the good file. The Master Playback Switch only existed in the good file. If it exists on the sound card and is muted by default, there would be no sound output. There are 3 possible duplicated controls. control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Surround Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Center Playback Switch' value true } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'LFE Playback Switch' value true } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Side Playback Switch' value.0 true value.1 true control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'CD Playback Switch' value.0 true value.1 true control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 true value.1 true control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Mic Playback Switch' value.0 true value.1 true control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true control.15 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true control.22 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true ----------------- The following controls existed in both files. Bad file, Good file Control 29 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Tone Control - Switch' value.0 false value.1 falseiface MIXER Bad file, Good file control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'IEC958 Optical Raw Playback Switch' value.0 false value.1 false Bad file, Good file NB also see control 13, an apparent duplicate control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Playback Switch' value true Bad file, Good file control.39 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Speaker Playback Switch' value true Bad file, Good file control.41 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value true Bad file, Good file NB also see control 9, a possible duplicate control.46 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Playback Switch' value true Bad file, Good file NB also see control 7, a possible duplicate. control.48 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'CD Playback Switch' value true Bad file, Good file control.52 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Aux Playback Switch' value true Bad file, Good file control.60 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name '3D Control - Switch' value false -- David Morrell Web site: www.davidmorrell.ozeweb.net (when I get it on air again :) <http://www.davidmorrell.ozeweb.net> Email: dsmorrell56(a)dodo.com.au <mailto:dsmorrell56@dodo.com.au> Ph: 0408 842 955 / 03 6343 5131

15 years, 8 months

Audigy soundcard silent after upgrade to Ubuntu Studio

by David Morrell

Hi all, my name is David, I'm new to the list. It was suggested that I try here by someone on the LAU list as they think I may have found a bug in the emu10k1 driver for Creative Audigy sound cards. Sound works in Windows XP and worked in Ubuntu 8.04. It stopped after a clean installation of Ubuntu Studio 9.04. It is not known if recording worked or if it is now broken. Hopefully sorting out playback will lead to a quick solution for recording, if one is needed. Summary of Test Results -------------------------------- Testing followed this understanding of the way sound playback is handled in Linux. Sound; - is generated by apps or test utilities such as Hydrogen or aplay - may pass through a sound server such as PulseAudio or Jack - goes to ALSA, which translates it for communication to sound cards - is converted from digital to analogue, amplified and fed to speakers by the sound card. The system has a Creative Labs Audigy2 sound card. It shares an IRQ with a video card and an onboard USB controller. Video and USB work properly in Linux. Video, USB and sound work in Windows. However, while BIOS reports these items on IRQ10, Linux (using cat /proc/interrupts) thinks they are on IRQ18. Is there any significance in this? It seems unlikely as video and USB work properly. ALSA appears to talk to the card – it is possible to set mixer controls using one mixer interface and read their state using an unrelated interface. Apps and test utilities appear able to send data to the card. To back this up, the kernel modules required to form the driver all appear to be loaded. When ALSA is forced to reload them, a pop and brief low level white noise comes from the speaker. This is similar to what can be heard during system boot. Possibly ALSA is passing control data to the card but not PCM audio data. I don't know if this is so or how to check. Re-installing ALSA had no effect. A full uninstall – reinstall was not done as many other packages would have been affected because of dependencies. Detailed information about the driver was reviewed, but it is beyond my technical competence to understand if or how it applies to my card and then to act on it (“Matrix:Module-emu10k1” at http://www.alsa-project.org/main/index.php/Matrix:Module-emu10k1). Now for the test results in detail. Thanks to Dave Phillips for his excellent series "Troubleshooting Linux Audio" at http://www.linuxjournal.com and to many other sources. Application & Test Utility Output -------------------------------- the Hyrdogen drum machine can be seen producing output on its VU meters. From the command line: aplay -vv ak4744.wav shows an apparently successful result; Playing WAVE 'ak4744.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'Audigy 2 Value [SB0400]' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 Test tone from Sound → Preferences → Test output appears to be successful to most sound pipelines; - Audigy 2 Multichannel Playback (ALSA) OK - Audigy 2 Multichannel Capture / PT Playback (ALSA) Fails with error message “gconfaudiosink: cound not open audio device for playback” (also describes the audio sources as a 512 Hz tone and the pipeline as “audioconvert ! Audioresample ! Gconfaudiosink”). - Audigy 2 ADC Capture / Standard PCM Playback (ALSA) OK - Audigy 2 ADC Capture / Standard PCM Playback (OSS) 2 are OK, the third apparently identical one fails - ALSA OK - OSS OK Sound Server -------------------------------- PulseAudio and JACK were both on the sytesm. To simplify the problem; JACK was not started; PulseAudio was completely uninstalled, the system was restarted and it was verified that no persistent processes still existed for PulseAudio (ps -e | grep “pulse”) Drivers (ALSA) -------------------------------- All volume controls were set to 100% and all mute controls were set to un-mute in Alsamixer. These settings were verified by having another mixer read and display them (Alsamixergui). As can be seen from the preceding output of aplay, sound was routed directly from its source to ALSA and ALSA recognised the Audigy 2 card on this computer. Output from the Sound → Preferences test also suggests that ALSA is interrogating the card and recognising what it can and cannot do. However, there is a strange indication from ALSCTL; alsactl init which shows; Unknown hardware: "Audigy2" "SigmaTel STAC9750,51" "AC97a:83847650" "" "" What is this saying? The ALSA soundcard matrix says an Audigy2 card should be using emu10k1 modules.. http://www.alsa-project.org/main/index.php/Matrix:Vendor-Creative_Labs though I can't find an exact match there for my Audigy 2 SB0400 card. I tried forcing ALSA to reload its kernel modules using; sudo alsa reload The speaker emitted a soft pop and about half a second of very low level white noise, similar to what can be heard during system boot. Commmand output was; lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/dmorrell/.gvfs Output information may be incomplete. /sbin/alsa: Warning: Processes using sound devices: 3215(timidity). Unloading ALSA sound driver modules: snd-emu10k1-synth snd-emux-synth snd-seq-virmidi snd-seq-midi-emul snd-emu10k1 snd-ac97-codec snd-pcm-oss snd-mixer-oss snd-pcm snd-page-alloc snd-util-mem snd-hwdep snd-seq-dummy snd-seq-oss snd-seq-midi snd-rawmidi snd-seq-midi-event snd-seq snd-timer snd-seq-device (failed: modules still loaded: snd-seq snd-timer snd-seq-device). Loading ALSA sound driver modules: snd-emu10k1-synth snd-emux-synth snd-seq-virmidi snd-seq-midi-emul snd-emu10k1 snd-ac97-codec snd-pcm-oss snd-mixer-oss snd-pcm snd-page-alloc snd-util-mem snd-hwdep snd-seq-dummy snd-seq-oss snd-seq-midi snd-rawmidi snd-seq-midi-event snd-seq snd-timer snd-seq-device. Kernel module(s) the card is using were checked using cat /proc/asound/modules which shows; 0 snd_emu10k1 i.e. there is only card 0 and it is using module snd_emu10k1. This was further confirmed using command lspci -v | grep “Audigy”, which showed; 05:09.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value Kernel driver in use: EMU10K1_Audigy A test was done to confirm that the modules are actually loaded into the kernel; lsmod | grep "emu" which showed a host of modules that seem as if they should be there, though of course could not show any modules that aren't there and should be; snd_emu10k1_synth 14336 0 snd_emux_synth 40832 1 snd_emu10k1_synth snd_seq_virmidi 13440 1 snd_emux_synth snd_seq_midi_emul 14592 1 snd_emux_synth snd_emu10k1 144288 1 snd_emu10k1_synth snd_ac97_codec 112292 1 snd_emu10k1 snd_pcm 83076 3 snd_emu10k1,snd_ac97_codec,snd_pcm_oss snd_page_alloc 16904 2 snd_emu10k1,snd_pcm snd_util_mem 12288 2 snd_emux_synth,snd_emu10k1 snd_hwdep 15108 2 snd_emux_synth,snd_emu10k1 snd_rawmidi 29696 3 snd_seq_virmidi,snd_emu10k1, snd_seq_midi snd_seq 56880 10 snd_emux_synth,snd_seq_virmidi,snd_seq_midi_emul,snd_seq_dummy, snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 29704 3 snd_emu10k1,snd_pcm,snd_seq snd_seq_device 14988 8 snd_emu10k1_synth,snd_emux_synth,snd_emu10k1,snd_seq_dummy,snd_seq_oss, snd_seq_midi,snd_rawmidi,snd_seq snd 62756 14 snd_emux_synth,snd_seq_virmidi,snd_emu10k1,snd_ac97_codec,snd_pcm_oss, snd_mixer_oss,snd_pcm,snd_hwdep,snd_seq_oss,snd_rawmidi,snd_seq,snd_ timer,snd_seq_device Just for the hell of it, I had a look at information about the emu10k1 module using; modinfo snd-emu10k1 which showed; filename: /lib/modules/2.6.28-15-generic/kernel/sound/pci/emu10k1/snd-emu10k1.ko license: GPL description: EMU10K1 author: Jaroslav Kysela <perex(a)perex.cz> firmware: emu/emu1010_notebook.fw firmware: emu/emu0404.fw firmware: emu/micro_dock.fw firmware: emu/emu1010b.fw firmware: emu/audio_dock.fw firmware: emu/hana.fw srcversion: F52CF37385CBD708CAB4A2C alias: pci:v00001102d00000008sv*sd*bc*sc*i* alias: pci:v00001102d00000004sv*sd*bc*sc*i* alias: pci:v00001102d00000002sv*sd*bc*sc*i* depends: snd-pcm,snd-util-mem,snd-page-alloc,snd,snd-rawmidi,snd-timer, snd-hwdep,snd-ac97-codec,snd-seq-device vermagic: 2.6.28-15-generic SMP mod_unload modversions 586 parm: index:Index value for the EMU10K1 soundcard. (array of int) parm: id:ID string for the EMU10K1 soundcard. (array of charp) parm: enable:Enable the EMU10K1 soundcard. (array of bool) parm: extin:Available external inputs for FX8010. Zero=default. (array of int) parm: extout:Available external outputs for FX8010. Zero=default. (array of int) parm: seq_ports:Allocated sequencer ports for internal synthesizer. (array of int) parm: max_synth_voices:Maximum number of voices for WaveTable. (array of int) parm: max_buffer_size:Maximum sample buffer size in MB. (array of int) parm: enable_ir:Enable IR. (array of bool) parm: subsystem:Force card subsystem model. (array of uint) Noticing the parameter 'enable' (ton enable a sound card) and feeling desparate, I tried reloading the module with that parameter specified; sudo modprobe -i snd_emu10k1 enable=true Unsurprisingly, it made no difference as the card already appears to be talking to its driver anyway. The ALSA soundcard matrix mentions that some cards may need the alsa-firmware package. That package is not on my system or in the Ubuntu repositories. However, its headers are installed. I don't know if this is significant. Next, I tried reinstalling ALSA. The following packages were affected; libsox-fmt-alsa alsa-base libasound2 libasound2-plugins mcp-plugins libesd-alsa0 This had no effect, so I considered completely uninstalling & reinstalling. However, many other packages that depend on ALSA would also have been removed. This looked like a short cut to a long reconstruction. Sound Card -------------------------------- The card works under Windows XP and had worked under Ubuntu 8.04. I have a Creative Labs Audigy 2 PCI card as card 0 on IRQ18 according to cat /proc/asound/cards 0 [Audigy2 ]: Audigy2 - Audigy 2 Value [SB0400] Audigy 2 Value [SB0400] (rev.0, serial:0x10011102) at 0x1040, irq 18 More information is given by cat /proc/interrupts which shows 18: 185146 IO-APIC-fasteoi uhci_hcd:usb4, EMU10K1, radeon@pci:0000:01:00.0 There is a USB controller, the soundcard (EMU10K1) and the video card (radeon) all using the same interrupt. The sound card does not appear on any other interrupts. BIOS reports all of these devices on IRQ10, not IRQ18. And yes, I have checked carefully to be sure I am not confusing 8 with 0. How could this discrepancy happen? A hint may lie in the online book “The Linux Kernel” at http://tldp.org/LDP/tlk/dd/pci.html. “For Intel based systems the system BIOS, which ran at boot time, has already fully configured the PCI system. This leaves Linux with little to do other than map that configuration. For non-Intel based systems further configuration needs to happen to: - Allocate PCI I/O and PCI Memory space to each device, - Configure the PCI I/O and PCI Memory address windows for each PCI-PCI bridge in the system, - Generate Interrupt Line values for the devices; these control interrupt handling for the device. “ Possibly Linux has read the information set up by the BIOS, then the PCI Fixup routine has remapped the Interrupt Line values even though.it may not have needed to do so? For a boots and all look at the PCI information about the sound card, lspci -vvxxx shows 05:09.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value Subsystem: Creative Labs Device 1001 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort - <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: I/O ports at 1040 [size=64] Capabilities: [dc] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-, D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: EMU10K1_Audigy Kernel modules: snd-emu10k1 The card has the following devices shown by; aplay -L This shows abridged); front:CARD=Audigy2,DEV=0 Audigy 2 Value [SB0400], ADC Capture/Standard PCM Playback Front speakers rear:CARD=Audigy2,DEV=0 Audigy 2 Value [SB0400], ADC Capture/Standard PCM Playback Rear speakers … Audigy 2 Value [SB0400], ADC Capture/Standard PCM Playback IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) To see all input devices; arecord -l which shows; arecord -l **** List of CAPTURE Hardware Devices **** card 0: Audigy2 [Audigy 2 Value [SB0400]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 1/1 card 0: Audigy2 [Audigy 2 Value [SB0400]], device 1: emu10k1 mic [Mic Capture] card 0: Audigy2 [Audigy 2 Value [SB0400]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 1/1 -- David Morrell Web site: www.davidmorrell.ozeweb.net (when I get it on air again :) <http://www.davidmorrell.ozeweb.net> Email: dsmorrell56(a)dodo.com.au <mailto:dsmorrell56@dodo.com.au> Ph: 0408 842 955 / 03 6343 5131 _______________________________________________ Linux-audio-user mailing list Linux-audio-user(a)lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user

15 years, 8 months

VamPy: Vamp plugins in Python

by Chris Cannam

VamPy, a Python wrapper for the Vamp plugin API, is now available. Using VamPy you can write audio analysis or visualisation plugins for use in Vamp hosts with a quick and dynamic environment that is somewhat like working in Matlab or other high-level modelling environments. VamPy has full two-way support for NumPy, an efficient numerical library for Python, and for the dynamic typing of Python. You can download VamPy from : http://www.vamp-plugins.org/vampy.html VamPy was written by Gyorgy Fazekas at the Centre for Digital Music, Queen Mary University of London and is published under a BSD-style license. Chris

15 years, 8 months

Regarding amplifying a PCM frame

by Drip Stone

Hi everybody, I have a question regarding amplifying a PCM frame. For each frame, I get a float number, and multiply this number by 2, and then output frames to alsa one by one using function snd_pcm_writei. But the wav file sounds the same to me, nothing gets amplified. Could anyone explain this to me? I appreciate your help! Warmest regards, Dripstone

15 years, 8 months

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Linux-audio-dev