Re: [Alsa-devel] An driver error when I using aplay!
Roc Wu wrote: # ./aplay -t wav -f U8 -r 22050 alarm.wav Playing WAVE 'alarm.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono ALSA lib pcm_plug.c:727:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' aplay: set_params:832: Sample format non available And the kernel driver will dump out the message as below: Badness in aaci_pcm_close at sound/arm/aaci.c:404 This seems to be a bug in the sound driver. However, arm/aaci.c is not part of the official ALSA distribution. AFAIK it's a patch written by Russell King. HTH Clemens --- This SF.Net email is sponsored by the new InstallShield X. From Windows to Linux, servers to mobile, InstallShield X is the one installation-authoring solution that does it all. Learn more and evaluate today! http://www.installshield.com/Dev2Dev/0504 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] limits
Florin Andrei wrote: What's the max number of cards in a system that can be used by ALSA simultaneously? 8 What's the max number of MIDI ports that's supported by ALSA? There can be up to 8 rawmidi devices per card, but each device can have an unlimited number of subdevices. OSS emulation supports 2 ports per card (/dev/midi0X, /dev/amidi0X). There can be up to 192 sequencer clients (64 each for kernel, drivers (8*8), and applications), and each can have up to 254 ports. What are other similar limits? Up to 8 playback and 8 capture PCM devices per card, with unlimited number of subdevices. Up to 8 sequencer queues. The answers to the questions above - are they in the docs? Use the Source, Luke! :-) HTH Clemens --- This SF.Net email is sponsored by the new InstallShield X. From Windows to Linux, servers to mobile, InstallShield X is the one installation-authoring solution that does it all. Learn more and evaluate today! http://www.installshield.com/Dev2Dev/0504 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] alsa-lib spamming terminal
Måns Rullgård wrote: Is there a way to make alsa-lib stop spamming the terminal with error messages, for instance when a non-blocking open fails? snd_lib_error_set_handler() HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id149alloc_id66op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] MAYA44 USB audio device doesn't work with 1.0.4.
Luke Yelavich wrote: usbcore: registered new driver snd-usb-audio No other message? There should be at least an error message when the driver fails to initialize. Please try to compile the kernel with ALSA's Debug detection enabled, or --with-debug=detect when configuring the package. Regards, Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Question regarding period/buffer and error handling
Cournapeau David wrote: My second problem is related to snd_card_next: If the card value returned by snd_card_next is different from -1, does that always mean than a alsa audio device is present on the system ? It means that same kind of device with an ALSA driver is present. This does not mean that you'll be able to play PCM data to it. And there may be software devices available. Is it the best method to probe a soundcard with alsa support? snd_card_next() and snd_ctl_pcm_next_device() will give you a list of hardware devices only. Usually, you should use the device named default and let the user allow to specify another device name. HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] MAYA44 USB audio device doesn't work with 1.0.4.
Luke Yelavich wrote: below is the information I get when attempting to use this soundcard under a 2.6.6 kernel with alsa drivers 1.0.5rc1 and hotplug. dmes: usbaudio: device 3 audiocontrol interface 0 has 1 input and 1 output AudioStreaming interfaces This is the OSS driver (audio). Put the line audio into the file /etc/hotplug/blacklist to prevent it from loading (or deactivate it in the kernel config). HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] RME 9632 Mixer...
Ed Wildgoose wrote: Can anyone suggest how to script the controls to default to some known values? alsactl store saves the current values of all mixer control in /etc/asound.state, alsactl restore restores them. HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] pcm buffer size
Nicolas Hüppelshäuser wrote: I'm using alsa 1.0.4. How can I get a larger alsa buffer size? PCM devices have a default maximum buffer size and a maximum maxmimum buffer size. For example, in the snd-usb-audio driver, it's 64 KB and 128 KB. The current value (in KB) can be seen in the file /proc/asound/cardX/pcmXY/subX/prealloc. You can change it with echo 128 /proc/asound/card0/pcm0p/sub0/prealloc In most cases, the maxmimum buffer size is constrained by hardware limitations, but in the case of snd-usb-audio, it should be safe to raise this value (look for the call to snd_pcm_lib_preallocate_pages() in usbaudio.c). HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id149alloc_id66op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1.0.5rc1 release
Please, report (especially compilation) problems. depmod: *** Unresolved symbols in /lib/modules/2.4.26/kernel/sound/isa/wavefront/snd-wavefront.o depmod: errno The open/close/read system calls are implemented as inline functions which change errno which isn't exported from the kernel. We could reintroduce the dummy definition of errno for older kernels, or use filp_* calls (like sound_firmware.c). Regards, Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] MPG123 and alsa
Tom Watson wrote:
I just downloaded a version (most recent I assume) of MPG123.
When I tried to do a 'make linux-alsa' I got a wholw bunch of
errors. While I suspect that this is because they are using an
older oversion of alsa (0.5.x?).
Yes.
and has anyone else done any mods to the mpg123 (or its library).
Try the patch below.
HTH
Clemens
--
diff -ur mpg123-pre0.59s-orig/audio.c mpg123-pre0.59s/audio.c
--- mpg123-pre0.59s-orig/audio.cThu Jan 18 15:00:33 2001
+++ mpg123-pre0.59s/audio.c Sat May 22 16:52:21 2004
@@ -17,9 +17,6 @@
ai-output = -1;
#ifdef ALSA
ai-handle = NULL;
- ai-alsa_format.format = -1;
- ai-alsa_format.rate = -1;
- ai-alsa_format.channels = -1;
#endif
ai-device = NULL;
ai-channels = -1;
diff -ur mpg123-pre0.59s-orig/audio.h mpg123-pre0.59s/audio.h
--- mpg123-pre0.59s-orig/audio.hThu Jan 18 15:00:33 2001
+++ mpg123-pre0.59s/audio.h Sat May 22 16:55:10 2004
@@ -42,7 +42,9 @@
#ifdef ALSA
-#include sys/asoundlib.h
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+#include alsa/asoundlib.h
#endif
struct audio_info_struct
@@ -59,7 +61,6 @@
int output;
#ifdef ALSA
snd_pcm_t *handle;
- snd_pcm_format_t alsa_format;
#endif
#ifdef AIX_UMS
UMSAudioDevice dev;
diff -ur mpg123-pre0.59s-orig/audio_alsa.c mpg123-pre0.59s/audio_alsa.c
--- mpg123-pre0.59s-orig/audio_alsa.c Mon Sep 20 16:55:02 1999
+++ mpg123-pre0.59s/audio_alsa.cSat May 22 21:20:31 2004
@@ -1,189 +1,168 @@
/*
- * Driver for Advanced Linux Sound Architecture, http://alsa.jcu.cz
+ * Driver for Advanced Linux Sound Architecture, http://www.alsa-project.org
*
* Code by Anders Semb Hermansen [EMAIL PROTECTED]
- * Cleanups by Jaroslav Kysela [EMAIL PROTECTED]
+ * Cleanups by Jaroslav Kysela [EMAIL PROTECTED]
* Ville Syrjala [EMAIL PROTECTED]
+ * Modified for ALSA 1.0.x by Clemens Ladisch [EMAIL PROTECTED]
*
- * You can use -a soundcard #:device #...
- * For example: mpg123 -a 1:0 aaa.mpg
- * mpg123 -a guspnp:1 aaa.mpg
+ * You can use -a devicename ...
+ * For example: mpg123 -a hw:1,0 aaa.mpg
+ * mpg123 -a guspnp aaa.mpg
*
* This file comes under GPL license.
*/
#include mpg123.h
-#include stdlib.h
-#include ctype.h
-#include sys/asoundlib.h
-
-#ifdef SND_LITTLE_ENDIAN
-#define SND_PCM_SFMT_S16_NE SND_PCM_SFMT_S16_LE
-#define SND_PCM_SFMT_U16_NE SND_PCM_SFMT_U16_LE
-#define SND_PCM_FMT_S16_NE SND_PCM_FMT_S16_LE
-#define SND_PCM_FMT_U16_NE SND_PCM_FMT_U16_LE
-#else
-#define SND_PCM_SFMT_S16_NE SND_PCM_SFMT_S16_BE
-#define SND_PCM_SFMT_U16_NE SND_PCM_SFMT_U16_BE
-#define SND_PCM_FMT_S16_NE SND_PCM_FMT_S16_BE
-#define SND_PCM_FMT_U16_NE SND_PCM_FMT_U16_BE
-#endif
-
int audio_open(struct audio_info_struct *ai)
{
int err;
- int card=0,device=0;
- char scard[128], sdevice[128];
+ const char *pcm_name;
if(!ai)
return -1;
- if(ai-device) {/* parse ALSA device name */
- if(strchr(ai-device,':')) {/* card with device */
- strncpy(scard, ai-device, sizeof(scard)-1);
- scard[sizeof(scard)-1] = '\0';
- if (strchr(scard,':')) *strchr(scard,':') = '\0';
- card = snd_card_name(scard);
- if (card 0) {
- fprintf(stderr, wrong soundcard number: %s\n, scard);
- exit(1);
- }
- strncpy(sdevice, strchr(ai-device, ':') + 1,
sizeof(sdevice)-1);
- } else {
- strncpy(sdevice, ai-device, sizeof(sdevice)-1);
- }
- sdevice[sizeof(sdevice)-1] = '\0';
- device = atoi(sdevice);
- if (!isdigit(sdevice[0]) || device 0 || device 31) {
- fprintf(stderr, wrong device number: %s\n, sdevice);
- exit(1);
- }
- }
-
- if((err=snd_pcm_open(ai-handle, card, device, SND_PCM_OPEN_PLAYBACK)) 0 )
- {
- fprintf(stderr, open failed: %s\n, snd_strerror(err));
- exit(1);
- }
- if(audio_reset_parameters(ai) 0)
+ pcm_name = ai-device ? ai-device : default;
+ if((err=snd_pcm_open(ai-handle, pcm_name, SND_PCM_STREAM_PLAYBACK, 0)) 0)
{
- audio_close(ai);
+ fprintf(stderr, cannot open %s: %s\n, pcm_name, snd_strerror(err));
return -1;
}
- return 0;
+ if(ai-format != -1)
+ /* we're going to play: initialize sample format */
+ return audio_reset_parameters(ai);
+ else
+ /* query mode; sample format will be set for each query */
+ return 0;
}
-static void audio_set_playback_params(struct audio_info_struct *ai)
+int audio_reset_parameters(struct audio_info_struct *ai)
{
int
Re: [Alsa-devel] 1.0.5rc1 release
Takashi Iwai wrote: I wrote: Clemens Ladisch wrote: depmod: *** Unresolved symbols in /lib/modules/2.4.26/kernel/sound/isa/wavefront/snd-wavefront.o depmod: errno The open/close/read system calls are implemented as inline functions which change errno which isn't exported from the kernel. does the attached patch work? +static int errno; I don't have access to my Linux machine here, but I guess it does. i guess sys_* functions are also not defined correctly on older kernels. they were open(), close(), read() with __KERNEL_SYSCALLS__. +#define sys_open open +#define sys_close close +#define sys_read read These definitions are already in alsa-driver/include/syscalls_26.h. I'm going to test and apply this until tomorrow, but feel free to apply it now if you want to release rc2 earlier. :) Regards, Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] direct midi event dispatching
Gilles Dégottex wrote: Why SND_SEQ_EVENT_NOTE event doesnt accept direct dispatching ? technicaly, there is surely a reason, but conceptualy, I dont see. Why do a difference ? SND_SEQ_EVENT_NOTE isn't a 'real' event but gets split into two events, SND_SEQ_EVENT_NOTE_ON and _OFF, and the latter must be sent at a later time. Scheduling events to be sent later requires a timer which isn't avilable without a queue. The SND_SEQ_EVENT_NOTE event is the only event that uses the event.data.note.duration field, but you cannot know what the value in this field means without having a timer which specifies how long one timer tick lasts. HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id149alloc_id66op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] USB audio kernel panic
Patrick Shirkey wrote:
Using the latest usb-audio code from cvs I get a kernel panic when
loading the drivers.
I'm running kernel-2.2.21 with gcc-2.95.4
Unable to handle kernel paging request at virtual address 28302e63
EIP:0010:[e08eafc0]Tainted: PF
e08e7060 __insmod_snd-usb-audio_S.text_L29104 [snd-usb-audio]
e08eb02c snd_hack_usb_set_interface [snd-usb-audio]
The 2.6.5-rc2 kernel introduced some API changes to work around
devices with broken descriptors. The resulting changes in the ALSA
code broke ALSA's workaround for the same problem.
The fix below is now in CVS.
HTH
Clemens
--
Index: usbaudio.inc1
===
RCS file: /cvsroot/alsa/alsa-driver/usb/usbaudio.inc1,v
retrieving revision 1.5
diff -u -r1.5 usbaudio.inc1
--- alsa-driver/usb/usbaudio.inc1 23 Mar 2004 12:48:18 - 1.5
+++ alsa-driver/usb/usbaudio.inc1 21 May 2004 07:34:10 -
@@ -98,7 +98,7 @@
interface, NULL, 0, HZ * 5)) 0)
return ret;
- iface-act_altsetting = alternate;
+ iface-act_altsetting = i;
for (i = 0; i iface_as-bNumEndpoints; i++) {
u8 ep = iface_as-endpoint[i].bEndpointAddress;
usb_settoggle(dev, epUSB_ENDPOINT_NUMBER_MASK, usb_endpoint_out(ep),
0);
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Re: [Alsa-devel] MAYA44 USB audio device doesn't work with 1.0.4.
Luke Yelavich wrote: As above. The vendor and product IDs given are 0xa92 and 0x91 respectively. If there is any more information I can give, please let me know. Output of lsusb -v? And what exactly does doesn't work mean? Regards, Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] mixer device
Ronald S. Bultje wrote:
is it possible for me to get the cardnumber (and thereby the hw:N
entry) for a given non-hw device (e.g. dspN, hw:N,M, dmix, ...)?
Try something like this:
/* Get the slave pcm of a pcm node. Caller must free the result. */
static snd_config_t *get_pcm_slave_pcm(snd_config_t *pcm)
{
snd_config_t *slave;
snd_config_t *result = NULL;
int free_slave = 0;
const char *str;
if (snd_config_search(pcm, slave, slave) 0)
goto _exit;
if (snd_config_get_string(slave, str) = 0) {
if (snd_config_search_definition(snd_config, pcm_slave, str, slave)
0)
goto _exit;
free_slave = 1;
}
if (snd_config_get_type(slave) != SND_CONFIG_TYPE_COMPOUND)
goto _exit;
if (snd_config_search(slave, pcm, pcm) 0)
goto _exit;
if (snd_config_get_string(pcm, str) = 0) {
if (snd_config_search_definition(snd_config, pcm, str, pcm) = 0)
result = pcm;
} else {
/* always make a dynamic copy, because the caller frees it */
if (snd_config_copy(pcm, pcm) = 0)
result = pcm;
}
_exit:
if (free_slave)
snd_config_delete(slave);
return result;
}
/* Get the card number from a pcm node. Searches recursively. */
static int get_pcm_card(snd_config_t *pcm)
{
snd_config_t *slave_pcm;
snd_config_t *card_node;
long card;
if (snd_config_get_type(pcm) != SND_CONFIG_TYPE_COMPOUND)
return -1;
slave_pcm = get_pcm_slave_pcm(pcm);
if (slave_pcm) {
card = get_pcm_card(slave_pcm);
snd_config_delete(slave_pcm);
return card;
} else if (snd_config_search(pcm, card, card_node) = 0) {
const char *str;
if (snd_config_get_integer(card_node, card) = 0)
return card;
if (snd_config_get_string(card_node, str) = 0)
return snd_card_get_index(str);
}
return -1;
}
/* Get the card number from a pcm device name. */
static int get_named_pcm_card(const char *pcm_name)
{
snd_config_t *pcm;
int card;
if (snd_config_update() 0)
return -1;
if (snd_config_search_definition(snd_config, pcm, pcm_name, pcm) 0)
return -1;
card = get_pcm_card(pcm);
snd_config_delete(pcm);
return card;
}
/* ... */
card = get_named_pcm_card(pcm_name);
if (card = 0)
sprintf(ctl_name, hw:CARD=%d, card);
else
/* unknown card */
HTH
Clemens
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Re: [Alsa-devel] BT878 support.
Rodrigo Severo wrote: With the Askey Magic I got only silence but with the Prolink PlayTV Pro revision 9D I got bad sound with btaudio. By bad I mean a slightly high pitch soundI got the sound of the TV with a slightly higher pitch than usual and with more than usual noise during silence. The sound was bad but at least I got something. Do you think there is any chance for these boards to work with btaudio and/or snd-bt87x? The driver supports the functionality that is implemented by the BT878 chip. If a card has some extra hardware for amplifying/routing/etc. that needs to be configured, the driver doesn't know about this. It would be possible to add this hardware-specific support if there would be some documentation about it. BTW, is there some kind of list with boards that work with btaudio and/or snd-bt87x? My Hauppauge WinTV works. It seems most other cards have added features that must be configured explicitly. Regards, Clemens --- This SF.Net email is sponsored by: SourceForge.net Broadband Sign-up now for SourceForge Broadband and get the fastest 6.0/768 connection for only $19.95/mo for the first 3 months! http://ads.osdn.com/?ad_id=2562alloc_id=6184op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] BT878 support.
Rodrigo Severo wrote: Clemens Ladisch wrote: Rodrigo Severo wrote: I have two Prolink PlayTV Pro capture cards (PV-BT878P+). One of them is revision 8E and tthe other is revision 9F. I tried to use both of them with the snd-bt87x driver. Both of them returned only static. Did you try analog or digital or both? I managed to get analog sound from the 8E with the btaudio (OSS) driver, Does digital work? Which version? ALSA 1.0.4 and btaudio which came with kernel 2.4.25. In theory, btaudio and snd-bt87x should behave identically. What btaudio mixer settings and input do you use when it works? Regards, Clemens --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] mixer device
Ronald S. Bultje wrote:
for a given alsa device (default, dmix, plughw:N, hw:N,M,
dspN, etc.), how do I automatically obtain the right device to use as
a audio mixer with that device?
Depends on your definition of right device. In the general case
this isn't possible because many cards don't have any mixer controls,
or don't have the controls you want to use. Or the pcm device uses
several cards, or none.
What do you want to do with the mixer?
So far, it seems I can only attach to hw:N mixer devices.
You can use any control device that is defined in your .asoundrc with
something like:
ctl.x {
type hw
card N
}
Regards,
Clemens
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Re: [Alsa-devel] ALSA snd-usb-audio kernel 2.6.5
Matt Flax wrote: The snd-usb-audio driver now doesn't hang my computer anymore during shutdown (it did with previous versions). Just BTW: Does it still work if you unload and then reload the snd-usb-audio module? :- Regards, Clemens --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] HELP! seq - writing to q in realtime mode
David McNab wrote: I'm trying to create and schedule outbound events, in realtime absolute mode. When sending these events directly, using SND_SEQ_QUEUE_DIRECT, the events get delivered fine to subscribers. But when adding realtime absolute timestamps, and marking the event to a queue I've created, only the very first event (time=0) gets sent. Is the queue running? (see /proc/asound/seq/queues) Can someone please give an overview (or preferably sample code) explaining what I need to do to transmit events in realtime absolute mode? See the aplaymidi example, which transmits events with relative tick timestamps. Set the appropriate bits in ev.flags (SND_SEQ_TIME_STAMP_REAL and SND_SEQ_TIME_MODE_ABS) and use ev.time.time.tv_sec and .tv_nsec instead of ev.time.tick. (The snd_seq_ev_schedule_real macro just sets these fields, too.) For instance, do I need to set up timers on the queue? No, queues have a default timer. HTH Clemens --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] USB audio kernel panic
Patrick Shirkey wrote: Using the latest usb-audio code from cvs I get a kernel panic when loading the drivers. I'm running kernel-2.2.21 with gcc-2.95.4 Works fine with my 2.2.19(?). ALSA 1.0.4 fixed some horrible bugs in the USB compatibility code for 2.2.x kernels. Where exactly does this crash happen? Regards, Clemens --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] spin_lock_irqwhat ?
Giuliano Pochini wrote: Some drivers use spin_lock_irq() a lot, while others always use spin_lock_irqsave(). I can't see the difference. When it's safe using the _irq() version ? Only when you know that you are not in interrupt context. Regards, Clemens --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] BT878 support.
Rodrigo Severo wrote: I have two Prolink PlayTV Pro capture cards (PV-BT878P+). One of them is revision 8E and tthe other is revision 9F. I tried to use both of them with the snd-bt87x driver. Both of them returned only static. Probably different mixer settings? I managed to get analog sound from the 8E with the btaudio (OSS) driver, Which version? Regards, Clemens --- This SF.Net email is sponsored by Sleepycat Software Learn developer strategies Cisco, Motorola, Ericsson Lucent use to deliver higher performing products faster, at low TCO. http://www.sleepycat.com/telcomwpreg.php?From=osdnemail3 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Samples: Float to Int conversions.
James Courtier-Dutton wrote: Which section of code in alsa-lib is doing these conversions. I would like to see what code you use for the task of converting samples in float format to samples in int format. Function snd_pcm_lfloat_convert_float_integer() in alsa-lib/src/pcm/pcm_lfloat.c. Regards, Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] ALI M5451 (CMI 9739): PCM always muted?
Nico Schottelius wrote: anyone knows if there is the possibility to adjust sound volume or not. CMI hardware doesn't have this capability. It would be possible to write a plugin that scales sound data in software before sending it to the device, but nobody has done this yet. Regards, Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Attempts on the ESS-1869 card (one more time)
Tom Watson wrote: Well, I've gone to another Compaq laptop. The previous one I had to resort to OSS (yuck) drivers and send it on its way. This laptop is one I'm going to use, so I'd like to get ALSA working on it. So... I've loaded ALSA and put the following in 'modules.conf': alias char-major-116 snd alias snd-card-0 snd-es18xx options snd-es18xx index=1 This is sound card 0, so use index=0. id=1 The id cannot be a number. But ALSA has a sensible default anyway, so you don't need to specify this. fm_port=0x338 fm_port=0x388 dma1=1 dma2=0 Does your card really use 1/0 and not 0/1? (To disable a dma channel, specify -1.) But alas, nothing works. What exactly doesn't work? Error messages when trying to load the driver? No sound? Green demons coming out of your speakers? ;-) HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] configuration space never flattened? Bug?
Erik Inge Bolsø wrote:
I'm having trouble doing snd_pcm_hw_params_can_pause() on a certain pcm,
even though it has already (hopefully?) been fixed to one config by
snd_pcm_hw_params()... snd_pcm_hw_params has been called, at least. How
can I check?
fileja {
type file
file /tmp/alsa
slave.pcm null
}
xine: pcm.c:2604: snd_pcm_hw_params_can_pause: Assertion `params params-info !=
~0U' failed.
Apparently, the null pcm doesn't initialize params-info.
The patch below should help.
HTH
Clemens
--
Index: alsa-lib/src/pcm/pcm_null.c
===
RCS file: /cvsroot/alsa/alsa-lib/src/pcm/pcm_null.c,v
retrieving revision 1.50
diff -u -r1.50 pcm_null.c
--- alsa-lib/src/pcm/pcm_null.c 17 Mar 2004 11:48:15 - 1.50
+++ alsa-lib/src/pcm/pcm_null.c 30 Apr 2004 09:38:09 -
@@ -276,6 +276,9 @@
{
int err = snd_pcm_hw_refine_soft(pcm, params);
params-fifo_size = 0;
+ params-info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_PAUSE;
return err;
}
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Re: [Alsa-devel] How do create a dummy device for recording audio? Does it exist already?
Caleb Crome wrote: Clemens Ladisch wrote: However, Realplayer uses the OSS emulation through /dev/dsp0. To make that work with your file device, add the following to the config file: pcm.dsp0 = file Awesome! It works like a charm. Than you so much! The one thing that doesn't work quite right is that I get the following error when RealPlayer attempts to access the mixer: ALSA lib control.c:601:(snd_ctl_open_noupdate) Invalid CTL mixer0 It tries to use /dev/mixer0, too. Add the following: ctl.mixer0 = hw:0 HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] [PATCH] intel8x0 - fixes 'periodical noisy output'
Arek Korbik wrote:
To summarise: hardware generates an interrupt when it should not - the
driver should handle it (the simplest solution: ignore). The attached
patch:
- adds interrupt-on-FIFOE handling(/ignoring) to handler routine
IMNSHO it's better to check for the buffer completion interrupt status
bit because that is the actual event we want to handle.
The patch below has been applied to CVS.
Regards,
Clemens
--
Index: intel8x0.c
===
RCS file: /cvsroot/alsa/alsa-kernel/pci/intel8x0.c,v
retrieving revision 1.136
diff -u -r1.136 intel8x0.c
--- intel8x0.c 21 Apr 2004 18:18:27 - 1.136
+++ intel8x0.c 26 Apr 2004 07:41:50 -
@@ -775,11 +775,14 @@
static inline void snd_intel8x0_update(intel8x0_t *chip, ichdev_t *ichdev)
{
unsigned long port = ichdev-reg_offset;
- int civ, i, step;
+ int status, civ, i, step;
int ack = 0;
+ status = igetbyte(chip, port + ichdev-roff_sr);
civ = igetbyte(chip, port + ICH_REG_OFF_CIV);
- if (civ == ichdev-civ) {
+ if (!(status ICH_BCIS)) {
+ step = 0;
+ } else if (civ == ichdev-civ) {
// snd_printd(civ same %d\n, civ);
step = 1;
ichdev-civ++;
@@ -813,7 +816,8 @@
snd_pcm_period_elapsed(ichdev-substream);
spin_lock(chip-reg_lock);
}
- iputbyte(chip, port + ichdev-roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
+ iputbyte(chip, port + ichdev-roff_sr,
+status (ICH_FIFOE | ICH_BCIS | ICH_LVBCI));
}
static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id, struct pt_regs *regs)
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Re: [Alsa-devel] [linux-audio-dev] DSP2000 C-PORT and MIDI input
Lukas Degener wrote: Hi everybody and sorry for posting this to both, lad and alsa-dev, i was not sure which one fits better. alsa-devel would be appropriate, but non-subscribers cannot post there, so I'm keeping linux-audio-dev in the CCs. Concerning the issue with non-functional midi in on the DSP2000 c-port: Someone on lau(Clemens?) said the problem is a lack of information an how to enable some interrupts. From what I've heard, interrupts just don't happen. The MIDI receiver is part of the ICE1712 chip, so, apparently, the MIDI input isn't enabled or in power saving mode or something like that. MIDI input works with this driver on ICE1712-based cards from other vendors, so this seems to be some Hoontech-specific configuration issue. Matthew Skinner, Sound and Music Tech Support wrote: If you can program in c++ and want to get the midi io working on the cport then let me know what you need. I need to know how exactly the box configuration registers(?) are to be initialized for DSP* cards. The driver currently fiddles with three bits named MIDIIN, MIDI1, and MIDI2, but their function seems to be more complex than the names imply. And the driver doesn't actually know about the C-Port. There are two known EEPROM subdevice IDs: 0x1712/0x1412 for the DSP24, and 0x1712/0x1416 for the Media7.1. I guess the C-Port uses one of these IDs? (Otherwise, this would explain why the C-Port isn't initialized completely.) Regards, Clemens --- This SF.net email is sponsored by: The Robotic Monkeys at ThinkGeek For a limited time only, get FREE Ground shipping on all orders of $35 or more. Hurry up and shop folks, this offer expires April 30th! http://www.thinkgeek.com/freeshipping/?cpg=12297 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] How do create a dummy device for recording audio? Does it exist already?
Caleb Crome wrote:
pcm.file {
type file
slave {
pcm hw:0,0
}
file /tmp/file.raw
}
Alright, I did what you suggested, and real player comes up and connects
to the dummy sound card (I had to do a modprobe snd-pcm-oss and
snd-mixer-oss). However, I don't get a file called /tmp/file.raw.
You have to configure the program to use the PCM named file instead
of the default default.
However, Realplayer uses the OSS emulation through /dev/dsp0. To make
that work with your file device, add the following to the config
file:
pcm.dsp0 = file
and try to run realplayer with the aoss script from the alsa-oss
package.
HTH
Clemens
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Re: [Alsa-devel] How to branch from WDM to ALSA fastest way??
ns wrote: I have WDM driver for our lab's own propertiary sound card(ADAT+SPDIF+DB...) and wanna write ALSA drv. I have linux w/kernel 2.4.20 (original). I know that I must patch kernel for ALSA and write ALSA driver. HOw to do it fastest? look into alsa-kernel/Documentation/DocBook/writing-an-alsa-driver HTH Clemens --- This SF.net email is sponsored by: The Robotic Monkeys at ThinkGeek For a limited time only, get FREE Ground shipping on all orders of $35 or more. Hurry up and shop folks, this offer expires April 30th! http://www.thinkgeek.com/freeshipping/?cpg=12297 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] How do create a dummy device for recording audio? Does it exist already?
Caleb Crome wrote:
Arek Korbik wrote:
pcm.file {
type file
slave {
pcm hw:0,0
}
file /tmp/file.raw
}
what do I do with that bit of configuration data? I found the
/usr/share/alsa/alsa.conf, and there is a similar pcm.file entry
in that, but it seems to take parameters.
Put it into your ~/.asoundrc file, or into /etc/asound.conf for all
users.
Since I have no sound card, how do I configure my system to use
the dummy PCM driver and use the pcm.file plugin?
modprobe snd-dummy should work.
HTH
Clemens
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Re: [Alsa-devel] midisport 8x8
ian esten wrote: On Wed, 2004-04-14 at 10:00, Clemens Ladisch wrote: What happens when you exchange the two 8x8 files in /usr/share/usb/midisport? i get a working midisport 8x8! i can send it midi, and it shows up in /proc/asound/cards. thanks so much! so why is it incorrectly being reported as hardware version 1.21? The reporting is correct, but I remembered that I wasn't sure which file was which when I wrote the extracting program. (This bug has now been fixed.) Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] alsa driver data rates
Brian Furey wrote: i'm using an open source VoIP application with the alsa driver. My card is the onboard intel8x0. My problem is figuring out the patterns I am getting with the alsa driver when transmitting packets. This depends more on how the VoIP application does things, not on the driver/soundcard. For instance when I use the GSM codec with a packets size of 80 msecs and 132 bytes, packets are transmitted at alternate intervals of 64/96 msecs. When i use G.711 packets of 240 bytes are transmitted every 32 msecs with an extra packet every 15 pkts. With G.711 it is obvious that 256 bytes are added to driver sending buffer every 32 msecs.However this does not match for GSM because it needs only 132 bytes every 64/96 msecs. Can the driver change the amount of data it gets from the soundcard to suit the codec? Have a look into /proc/asound/card0/pcm0?/sub0/hw_params to see the period size. The period size is constant, but if it is small enough the application can just wait until enough periods have accumulated for one packet. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] midisport 8x8
ian esten wrote: the midisport i have is reported as being hardware version 1.33 by the midisport control panel under windows xp (where it works fine). i looked in the midisport_fw loader, and it seems to only know about hardware versions 1.10 and 1.21. here are the entries in /var/log/messages when it is powered up: Apr 13 14:49:33 [/etc/hotplug/usb/midisport_fw] load /usr/share/usb/midisport/MidiSport8x8-2.21.ihx for 763/1031/121 to /proc/bus/usb/002/009 21 hexadecimal = 33 decimal Apr 13 14:49:33 [kernel] usb.c: USB disconnect on device 00:1f.4-1 address 9 Apr 13 14:49:34 [kernel] hub.c: new USB device 00:1f.4-1, assigned address 10 Apr 13 14:49:39 [kernel] usb_control/bulk_msg: timeout - Last output repeated 4 times - Apr 13 14:50:00 [kernel] hub.c: new USB device 00:1f.4-1, assigned address 11 Apr 13 14:50:05 [kernel] usb_control/bulk_msg: timeout the midisport never shows up in /proc/asound/cards. I guess it doesn't shop up in /proc/bus/usb/devices or in the output of lsusb, either? What version of the firmware loader do you use? What happens when you exchange the two 8x8 files in /usr/share/usb/midisport? i also have a midisport 8x8 at work (hardware version 1.21) What hexadecimal version number does this one have? (bcdDevice field of lsusb -v) which i brought home to check i had everything set up ok. here is what it reports to /var/log/messages: Apr 13 14:53:12 [kernel] hub.c: connect-debounce failed, port 1 disabled Apr 13 14:53:14 [kernel] hub.c: new USB device 00:1f.4-1, assigned address 12 Apr 13 14:53:14 [kernel] devfs_register(unknown): could not append to parent, err: -17 Apr 13 14:53:17 [usb.agent] missing kernel or user mode driver snd-usb-audio Are these all the messages for this device? If yes, then this one already has the firmware in its ROM and doesn't need a firmware download. Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Adding a Mute Switch to snd-tvmixer
Perry Gilfillan wrote: I've a couple of people using my snd-tvmixer module with success now, and would appreciate comments/critcism/abuse (any thing is better than the deathly silence my first announcement yeilded) from people more experienced than myself. Looks nice. (Not that I actually have used it :) In particular, I'd like to know what I may have missed in getting a Mute Switch to show up in gnome-alsa-mixer. Does it show up in alsamixer? In /etc/asound.state after running alsactl store? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] DigiDesign MBox USB Audio device
Patrick Boettcher wrote: I only applied the patch, loaded the module, pluged in the device ... result is attached. EIP is at __wake_up_common+0x13/0x50 Call Trace: [c0119a0c] __wake_up+0x32/0x57 [c015cc1e] pipe_readv+0x288/0x2d8 [c015cca5] pipe_read+0x37/0x3b [c0150d74] vfs_read+0xd0/0x135 [c0151007] sys_read+0x42/0x63 [c01071ef] syscall_call+0x7/0xb note: sed[5839] exited with preempt_count 1 EIP is at fasync_helper+0x39/0xed Call Trace: [c015d244] pipe_read_fasync+0x48/0x68 [c015d36b] pipe_read_release+0x0/0x3f [c015d38a] pipe_read_release+0x1f/0x3f [c0151e50] __fput+0x116/0x128 [c0150562] filp_close+0x59/0x86 [c01505f2] sys_close+0x63/0x96 [c01071ef] syscall_call+0x7/0xb note: usb.agent[5828] exited with preempt_count 1 Strange. But having two endpoints in one interface is not what the driver was written for. I think part of the driver needs to be rewritten before this device can work. Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] [PATCH] new options for arecordmidi
Pedro Lopez-Cabanillas wrote: This patch adds three new options to arecordmidi. Committed, with some changes. Thanks, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Bug in sis7012 driver - or probably I'm doing something wrong ?
Robert Rozman wrote:
I've got advice how to use this 6 channel device as 3 stereo channels but
there seems something not to be right (it could be my mistake).
I have /etc/asound.conf :
pcm.a {
type plug
slave.pcm ch12
}
pcm.b {
type plug
slave.pcm ch34
}
Now I do:
- alsaplayer with mp3 file on device plug:a
a is already defined as a plug plugin on top of ch12, it's exactly
the same as plug:ch12. Adding another plug layer (plug:plug:ch12)
wouldn't make much sense.
- aplay -D plug:ch12 (funny is that plug:a doesn't work)
- everything is ok and mixed properly
- but if I start another alsaplayer on plug:b (probably also plug:c would be
the same), then aplay produces sound that is interleaved in short chunks
and also sound from alsaplayer on plug:b is also heard in addition to
original mp3 and aplay sound.
Please try the a,b,c pcm devices without the plug: prefix.
Does it work if you run aplay on devices a and b?
Regards,
Clemens
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Re: [Alsa-devel] snd_pcm_writei not returning (?!?)
Tom Browne wrote: Clemens Ladisch wrote: Tom Browne wrote: Under certain conditions (yet to be established properly - it doesn't happen every single time) with this ICE1712 card (DSP24), snd_pcm_writei doesn't just block... it never comes back. What is the state of the PCM device when this happens? (see /proc/asound/cardX/pcm0p/*) state: PREPARED The pcm isn't (yet) running. avail : 5 And the buffer is almost full. snd_pcm_writei blocks because it cannot write more data to the buffer, probably because the transfer threshold is too high. How do you start the pcm? If you're using the start threshold, it may be too big, too. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cannot submit datapipe for urb 0 on /dev/dsp access -extigy
John H. wrote: the usb 2.0 hub was at fault,. thanks! I have heard rumors that the latest 2.6.5 kernel is supposed to have support for playing audio data through 2.0 hubs to 1.x devices. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] DigiDesign MBox USB Audio device
Patrick Boettcher wrote: On Mon, 5 Apr 2004, Clemens Ladisch wrote: What sample rate/format uses 304000 bytes per second? I don't understand the question completely, what do you mean? The descriptors say that the device can transfer up to 304000 bytes per second in each direction. What sample rate and formats does it support? (stereo 24 bits at 48 kHz would be 288000 bytes/s) 2. Can I use somehow the i2c/cs8427.c? Probably not. The CS8427 will be connected (via I2C) to some USB interface/controller chip. Can you determine which one? TAS1020A, Texas Instrument USB Streaming Controller. I thought I have to talk to the sound chip directly and that the the usb chip is only for bus control?! The USB chip manages communication on the bus, and contains an embedded 8051(?)-compatible CPU which must be programmed to implement the actual behaviour of the device (in this case, writing/reading samples over the I2C bus). Beside of the a/d and d/a converter, there is only one bigger chip on the device board. It is an EEPROM, I suppose to take the firmware, the owner of the device told me, that sometimes he has to make a firmware update. What exactly does the firmware? It's the code running on the TAS1020A's CPU. Sending/receiving sample data should work, but I guess this device needs vendor-specific commands to configure it. These are control transfers, not isochronous; please look in the log file for them. There are only isochronous transfers (of course, there are only isoc-endpoints?) visible. Each interface has an implicit control endpoint. There are some control transfers in the UsbSnoop log, but those look very much like standard USB Audio request. It seems the device only pretends to be vendor-specific. I think I'll write a quirk for this ... Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] problem while opening device
Pavana Sharma wrote: ALSA lib pcm.c:1947:(snd_pcm_open_noupdate) Unknown PCM hw:0,0 Please show the output of aplay -l and aplay -L. Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] DigiDesign MBox USB Audio device
I wrote:
I think I'll write a quirk for this ...
Please try the attached patch.
HTH
Clemens
diff -urN alsaold/alsa-kernel/usb/usbaudio.c alsa/alsa-kernel/usb/usbaudio.c
--- alsaold/alsa-kernel/usb/usbaudio.c Tue Mar 30 13:51:30 2004
+++ alsa/alsa-kernel/usb/usbaudio.c Tue Apr 6 13:20:36 2004
@@ -2640,6 +2640,7 @@
struct audioformat *fp;
struct usb_host_interface *alts;
int stream, err;
+ int *rate_table = NULL;
fp = kmalloc(sizeof(*fp), GFP_KERNEL);
if (! fp) {
@@ -2647,16 +2648,30 @@
return -ENOMEM;
}
memcpy(fp, quirk-data, sizeof(*fp));
+ if (fp-nr_rates 0) {
+ rate_table = kmalloc(sizeof(int) * fp-nr_rates, GFP_KERNEL);
+ if (!rate_table) {
+ kfree(fp);
+ return -ENOMEM;
+ }
+ memcpy(rate_table, fp-rate_table, sizeof(int) * fp-nr_rates);
+ fp-rate_table = rate_table;
+ }
+
stream = (fp-endpoint USB_DIR_IN)
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
err = add_audio_endpoint(chip, stream, fp);
if (err 0) {
kfree(fp);
+ if (rate_table)
+ kfree(rate_table);
return err;
}
if (fp-iface != get_iface_desc(iface-altsetting[0])-bInterfaceNumber ||
fp-altset_idx = iface-num_altsetting) {
kfree(fp);
+ if (rate_table)
+ kfree(rate_table);
return -EINVAL;
}
alts = iface-altsetting[fp-altset_idx];
diff -urN alsaold/alsa-kernel/usb/usbquirks.h alsa/alsa-kernel/usb/usbquirks.h
--- alsaold/alsa-kernel/usb/usbquirks.h Thu Mar 18 21:45:03 2004
+++ alsa/alsa-kernel/usb/usbquirks.hTue Apr 6 13:44:07 2004
@@ -838,6 +886,72 @@
.product_name = PHASE 26,
.ifnum = 3,
.type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+
+{
+ USB_DEVICE(0x0dba, 0x1000),
+ .driver_info = (unsigned long) (const snd_usb_audio_quirk_t) {
+ .vendor_name = Digidesign,
+ .product_name = MBox,
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ /*
+* The MBox has both endpoints in interface 1,
+* but we need one interface for each
+* direction, so use interface 0 here.
+*/
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3BE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = EP_CS_ATTR_SAMPLE_RATE,
+ .endpoint = 0x81,
+ .ep_attr = EP_ATTR_ASYNC | 0x01,
+ .maxpacksize = 0x130,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .nr_rates = 2,
+ .rate_table = (unsigned int[]) {
+ 44100, 48000
+ }
+ }
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3BE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = EP_CS_ATTR_SAMPLE_RATE,
+ .endpoint = 0x02,
+ .ep_attr = 0x01,
+ .maxpacksize = 0x130,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max =
Re: [Alsa-devel] snd_pcm_writei not returning (?!?)
Tom Browne wrote: Under certain conditions (yet to be established properly - it doesn't happen every single time) with this ICE1712 card (DSP24), snd_pcm_writei doesn't just block... it never comes back. What is the state of the PCM device when this happens? (see /proc/asound/cardX/pcm0p/*) Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] DigiDesign MBox USB Audio device
Patrick Boettcher wrote: I'm using a usbsnoop program to do some basic sniffing. Logs are really big :/. Try to play some 0.01-seconds wave file, and don't connect any other USB devices. After plugging in, the device shows like this in /proc/bus/usb/devices: I: If#= 0 Alt= 0 #EPs= 0 Cls=ff(vend.) Sub=00 Prot=00 Driver=(none) I: If#= 1 Alt= 0 #EPs= 0 Cls=ff(vend.) Sub=00 Prot=00 Driver=(none) I: If#= 1 Alt= 1 #EPs= 2 Cls=ff(vend.) Sub=00 Prot=00 Driver=(none) E: Ad=81(I) Atr=05(Isoc) MxPS= 304 Ivl=1ms E: Ad=02(O) Atr=05(Isoc) MxPS= 304 Ivl=1ms Class 0xff means vendor-specific. This is not a USB Audio Device as defined in the USB specifications. What sample rate/format uses 304000 bytes per second? After talking to some more experienced guys, I know, that there has to be a sound chip, so I opened the device and among some A/D and D/A converter I found a cs8427 chip. I suppose this is the one. After looking into sound directory in linux, I saw i2c/cs8427.c and i2c/cs8427.h. 1. How should a go on? Output of lsusb -v? 2. Can I use somehow the i2c/cs8427.c? Probably not. The CS8427 will be connected (via I2C) to some USB interface/controller chip. Can you determine which one? 3. How does communication with the device should work? Sending/receiving sample data should work, but I guess this device needs vendor-specific commands to configure it. These are control transfers, not isochronous; please look in the log file for them. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cannot submit datapipe for urb 0 on /dev/dsp access -extigy
John H. wrote: see, this is what is weird, it worked last week, now it doesn't work with exaudio or alsa! same errors! Then it's not a problem of the sound driver. cannot submit datapipe for urb 0, err = -38 -38 means not implemented. Probably some bug in the USB core in your kernel. Have you connected the device through a USB 2.0 hub? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Disabling onboard codec
Ron Cococcia wrote: In a setup I'm working with, I have a motherboard (815 w/82801BA-ICH2) that has an onboard/primary (00) AC97 codec (AD1885). We've been playing around with adding a CNR card that would have a different/secondary (01) codec on it (CS4299). [...] The onboard chip is generating (I think) a BIT_CLK signal, and (I think) the CNR is as well (it's wired to an oscillator). There is exactly one clock signal on the AC'97 bus, driven either by the primary codec or (more often) by the controller. I guess the oscillator on the CMR card is for a configuration where the CS4299 would be the primary codec. I'm worried that both chips generating a BIT_CLK might cause jitter, and could cause audio quality issues. Even worse, two different clocks would drift apart over time, which would make it impossible to send synchronized samples to both codecs. Having two independent clocks in an audio system would require resampling. The second codec in an AC'97 system should not affect the primary one. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cannot submit datapipe for urb 0 on /dev/dsp access -extigy
John H. wrote: I get this error repeatedly ALSA /tmp/alsa-driver-1.0.4rc2/usb/usbaudio.c:811: cannot submit datapipe for urb 0, err = -38 whenever i try to use /dev/dsp now, and i can only get sound from line in. i have kernel 2.6.4 and drivers 1.0.4rc2 What is in /proc/asound/cards? Is your host controller UHCI or OHCI? Does it work with aplay something.wav? Does it work with a 2.4.x kernel? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1.0.4rc1 release
Jaroslav Kysela wrote: On Wed, 31 Mar 2004, Clemens Ladisch wrote: But now: # modprobe snd modprobe: Too deep recursion in module dependencies! modprobe: Circular dependency? snd-page-alloc snd snd_pci_compat_get_pci_driver() is called from pci_compat_pm_callback() in misc.c. I don't see the circular dependency for this function. Sorry, I didn't remember that snd-page-alloc should not depend on snd. The circular dependency is caused by snd_pci_compat_request_region() which calls request_region() which is #defined to be snd_wrapper_request_region() in wrappers.c. snd_pci_compat_request_region() clearly didn't intend to use the wrapper function, so I added another wrapper function to call the original function. Now the modules load. However, loading snd-usb-audio doesn't do anything, and unloading and reloading it results in a crash or a hang. Still debugging ... Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1.0.4rc1 release
I wrote: Now the modules load. However, loading snd-usb-audio doesn't do anything, and unloading and reloading it results in a crash or a hang. Fixed. Apparently, snd-usb-audio has never been working on 2.2.x, and nobody noticed ... No remaining 2.2.x bugs AFAICS. Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1.0.4rc1 release
Jaroslav Kysela wrote: 1.0.4rc1 packages are out. Please, report compilation problems. On a 2.2.19 kernel: depmod: *** Unresolved symbols in /lib/modules/2.2.19/misc/snd-* depmod: synchronize_irq depmod: tqueue_lock depmod: waitqueue_lock depmod: *** Unresolved symbols in /lib/modules/2.2.19/misc/snd-au88?0.o depmod: snd_pci_compat_disable_device depmod: *** Unresolved symbols in /lib/modules/2.2.19/misc/snd.o depmod: kernel_flag How long do we want to support 2.2.x? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] 1.0.4rc1 release
Jaroslav Kysela wrote: On Tue, 30 Mar 2004, Clemens Ladisch wrote: On a 2.2.19 kernel: depmod: *** Unresolved symbols in /lib/modules/2.2.19/misc/snd-* ... Missing inclusion of some header files? Sorry, my error. Wrong kernel configuration. (The result of Hey, this old file looks like a kernel -- let's see what happens if I boot it ... ;-) But now: # modprobe snd modprobe: Too deep recursion in module dependencies! modprobe: Circular dependency? snd-page-alloc snd snd_pci_compat_get_pci_driver() is called from pci_compat_pm_callback() in misc.c. I think the 2.2.x CONFIG_APM stuff in misc.c could be moved into pci_compat_22.c, but OTOH it might make more sense to move the PCI functions that don't do page allocation from the snd-page-alloc module into snd. What's your opinion on this? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] help needed-- providing control interface
Pavana Sharma wrote: I am providing control interface for my driver to access from user space with functions get put . * while creating new controls should I be allowed to create my hardware specific controls like gain, power, rate other codec specific parameters. All controls are hardware-specific. There are no requirements what controls you must, or must not, provide. or should I take care of mapping OSS mixer controls to ALSA mixer. ALSA does this automatically, based on the control names. * How user comes to know what controls are available the valid range of values for sliders switches. The driver defines type and range of a control in the info callback. The user can determine these things with the snd_ctl_* functions in the ALSA library. See the source code of amixer and alsactl for details. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd_pcm_wait() More details required.
Jaroslav Kysela wrote: On Tue, 23 Mar 2004, Clemens Ladisch wrote: The following test program shows strange behaviour: Thanks. It's fixed in CVS now. The second problem persists, i.e., snd_pcm_wait() returns although the pcm isn't actually ready: $ ./waittest dmix:hw:0 12000 frames written pcm is not ready! 6000 frames written 6000 frames written 6000 frames written ... Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd_pcm_wait() More details required.
Jaroslav Kysela wrote:
On Mon, 22 Mar 2004, James Courtier-Dutton wrote:
My current tests were not using the resampler. i.e. app using 48k, sound
card using 48k. I will try some tests that would use the resampler.
Then I need more info how I can reproduce the bug.
The following test program shows strange behaviour:
// -- bite off here
#include alsa/asoundlib.h
#define CHECK(command) do { \
err = (command); \
if (err 0) { \
printf(%s failed: %s\n, #command, snd_strerror(err)); \
return 1; \
} \
} while (0)
int main(int argc, char *argv[])
{
snd_pcm_t *pcm;
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
int err;
static char silence[12000 * 2 * 2];
if (argc 2) return 1;
CHECK(snd_pcm_open(pcm, argv[1], SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK));
snd_pcm_hw_params_alloca(hw_params);
CHECK(snd_pcm_hw_params_any(pcm, hw_params));
CHECK(snd_pcm_hw_params_set_access(pcm, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED));
CHECK(snd_pcm_hw_params_set_format(pcm, hw_params, SND_PCM_FORMAT_S16));
CHECK(snd_pcm_hw_params_set_channels(pcm, hw_params, 2));
CHECK(snd_pcm_hw_params_set_rate(pcm, hw_params, 48000, 0));
CHECK(snd_pcm_hw_params_set_buffer_size(pcm, hw_params, 12000));
CHECK(snd_pcm_hw_params_set_period_size(pcm, hw_params, 6000, 0));
CHECK(snd_pcm_hw_params(pcm, hw_params));
snd_pcm_sw_params_alloca(sw_params);
CHECK(snd_pcm_sw_params_current(pcm, sw_params));
CHECK(snd_pcm_sw_params_set_start_threshold(pcm, sw_params, 12000));
CHECK(snd_pcm_sw_params_set_avail_min(pcm, sw_params, 6000));
CHECK(snd_pcm_sw_params(pcm, sw_params));
for (;;) {
CHECK(snd_pcm_wait(pcm, 2000));
if (err == 0)
printf(timeout!\n);
err = snd_pcm_writei(pcm, silence, 12000);
if (err = 0)
printf(%d frames written\n, err);
else if (err == -EAGAIN)
printf(pcm not ready!\n);
else
CHECK(err);
}
}
// -- bite off here
With an AC'97 device:
$ ./waittest hw:0
12000 frames written
6000 frames written
6000 frames written
6000 frames written
6000 frames written
...
$ ./waittest dmix:hw:0
timeout!
12000 frames written
pcm is not ready!
6000 frames written
6000 frames written
6000 frames written
6000 frames written
...
When running with dmix, the first call to snd_pcm_wait() times out
although the pcm is ready (it seems the hardware pcm is still stopped
and doesn't generate interrupts), and the second call to
snd_pcm_wait() returns 1 although the pcm is not yet ready.
Regards,
Clemens
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Re: [Alsa-devel] snd-usb-audio and midisport1x1 little hack fixes hotplug pb
Hartmut Geissbauer wrote: I removed the quick hack from Mathieu and inserted snd_printk's in snd_usbmidi_disconnect. The new function: ... if (ep-out ep-out-urb) usb_unlink_urb(ep-out-urb); snd_printk(KERN_INFO usb_unlink_urb out\n); Please note that this is C, not Python. :-) The output from dmesg I got: ohci_hcd :00:02.0: urb e2e2b4c0 path 2.4 ep1in 5e16 cc 5 -- status -110 hub 1-2:1.0: port 4, status 100, change 1, 12 Mb/s usb 1-2.4: USB disconnect, address 5 usb 1-2.4: usb_disable_device nuking all URBs It seems we don't need to unlink the URBs as the kernel does this for us, but it doesn't hurt. ... ALSA sound/usb/usbmidi.c:723: usb_unlink_urb in ALSA sound/usb/usbaudio.c:2924: snd_usb_audio_disconnect called, refcount = 0 Apparently, snd_usb_audio_disconnect is called recursively, caused by the call to usb_driver_release_interface, and then deadlocks. The USB core will release all interfaces automatically, so we don't need this call. Please remove the usb_driver_release_interface line. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Opening dev/dsp takes very long
Lars Heineken wrote: Takashi Iwai wrote: | try strace with -r option and check which call takes too long time. When I use the application 'play' the delay is only around one second, the strace output is attached below. It seems play is a shell script that calls sox to play the file. Please try to strace sox. Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd-usb-audio and midisport1x1 little hack fixes hotplug pb
Mathieu Geli wrote:
Ok, I applied your second patch, and get this dmesg output:
drivers/usb/core/usb.c: deregistering driver snd-usb-audio
ALSA /home/mathieu/alsa-driver/usb/usbaudio.c:2944: snd_usb_audio_disconnect called,
refcount = 1
ALSA /home/mathieu/alsa-driver/usb/usbaudio.c:2944: snd_usb_audio_disconnect called,
refcount = 0
Two calls to disconnect? Strange.
ALSA /home/mathieu/alsa-driver/alsa-kernel/usb/usbmidi.c:148: urb status -104
ALSA /home/mathieu/alsa-driver/alsa-kernel/usb/usbmidi.c:134: usb_submit_urb: -32
The driver tries to resubmit the URB although the device is being shut
down. I don't think this causes the hang, but the patch below
(already in CVS) fixes this. (-104 is -ECONNRESET which shouldn't
occur when unlinking synchronously.)
Apparently, snd_usb_audio_disconnect hangs before returning. Please
try to insert printk's in snd_usbmidi_disconnect as Takashi suggestes.
Regards,
Clemens
--
diff -u -r1.25 -r1.26
--- alsa-kernel/usb/usbmidi.c 8 Mar 2004 09:34:05 - 1.25
+++ alsa-kernel/usb/usbmidi.c 16 Mar 2004 16:36:05 - 1.26
@@ -143,8 +143,9 @@
if (status == -ENOENT)
return status; /* killed */
if (status == -EILSEQ ||
+ status == -ECONNRESET ||
status == -ETIMEDOUT)
- return -ENODEV; /* device removed */
+ return -ENODEV; /* device removed/shutdown */
snd_printk(KERN_ERR urb status %d\n, status);
return 0; /* continue */
}
@@ -706,7 +707,6 @@
int i;
umidi = list_entry(p, snd_usb_midi_t, list);
- usb_driver_release_interface(driver, umidi-iface);
for (i = 0; i MIDI_MAX_ENDPOINTS; ++i) {
snd_usb_midi_endpoint_t* ep = umidi-endpoints[i];
if (ep-out ep-out-urb)
@@ -714,6 +714,7 @@
if (ep-in ep-in-urb)
usb_unlink_urb(ep-in-urb);
}
+ usb_driver_release_interface(driver, umidi-iface);
}
static void snd_usbmidi_rawmidi_free(snd_rawmidi_t* rmidi)
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Re: [Alsa-devel] the emu10k1 midi bug
Ove Kaaven wrote: - wavetable drops notes sometimes. (a good test example wanted.) I've encountered this randomly sometimes, but I think I also have an example of the opposite - a note that is never turned off - that manifests every single time for me. So I'm wondering if anyone else can reproduce. 1. Install MusE 0.6.3. (pmidi or aplaymidi does not exhibit the problem, but MusE does; If it doesn't happen with aplaymidi, it's probably not an emu10k1 bug. Did you try playmidi? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd-usb-audio hang when rmmoding after unplugging midisport1x1
Mathieu Geli wrote:
ok, what I did, is first to apply your patch to my source tree
(1.0.3rc2), clean, compile, install, and even reboot. That doesn't
output anymore the two lines urb status -104, and
usb_submit_urb: -32 but still hang after printing:
drivers/usb/core/usb.c: deregistering driver snd-usb-audio
It seems to work fine on my 2.4.25 kernel.
Can you unload the snd-usb-audio module? If not, please apply the
patch below to see whether the driver or the USB core is the culprit.
Regards,
Clemens
--
--- alsa-kernel/usb/usbaudio.c 8 Mar 2004 09:29:51 - 1.87
+++ alsa-kernel/usb/usbaudio.c 12 Mar 2004 12:05:22 -
@@ -2928,6 +2928,7 @@
return;
chip = snd_magic_cast(snd_usb_audio_t, ptr, return);
+ snd_printk(KERN_INFO snd_usb_audio_disconnect called, refcount = %d\n,
chip-num_interfaces);
card = chip-card;
down(register_mutex);
chip-shutdown = 1;
@@ -2947,6 +2948,7 @@
} else {
up(register_mutex);
}
+ snd_printk(KERN_INFO snd_usb_audio_disconnect finished!\n);
}
/*
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Re: [Alsa-devel] USB audio devices
Karsten Wiese wrote: We can also vary the exact USB frame time. With UHCI 1.1 USB Hosts there is the SOF Register. ... It really works here already with the us428: The trick is: We first make the USB-Frame longer until we capture 1 Sample Frame more 45 (for 44100). then the USB-Frame is shortened until we capture only 43 Sample Frames for one USB Frame. and so on. Cool idea! However, standard-compliant devices are supposed to base their timing for feedback/feedforward information on the arrival of SOF packets. Introducing jitter into the SOF timer may not be a good idea in every case. And this won't work if there is more than one audio device connected to the same bus, or if the HC is OHCI or EHCI. Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] USB audio devices
Clemens Ladisch wrote: Jaroslav Kysela wrote: I don't know much about USB 2.0, Not too much differences for the driver. Well, I should have read the specification before saying such things. The format of synchronization information has changed, too; it's not 10.14 bits but 12.13 packed into 16.16 (4 bytes). Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] snd-usb-audio hang when rmmoding after unplugging midisport1x1
Mathieu Geli wrote:
I'm trying to plug/unplug the midisport1x1 from M-Audio. The plug
works like a charm, hotplug does its job, load the firmware
(fxload) and snd-usb-audio, if not inserted, is auto-loaded. But
hen I unplug the device, and I want to plug it again, the usb
just hang, trying to rmmoding snd-usb-audio or anything else usb
related will just hang.
There was a bug which has been fixed three days ago (after 1.0.3 came
out). Either get the latest version from the CVS, or apply the patch
below.
HTH
Clemens
--
--- alsa-kernel/usb/usbmidi.c 3 Mar 2004 11:35:15 - 1.24
+++ alsa-kernel/usb/usbmidi.c 8 Mar 2004 09:34:05 - 1.25
@@ -453,8 +453,16 @@
usbmidi_out_port_t* port =
(usbmidi_out_port_t*)substream-runtime-private_data;
port-active = up;
- if (up)
+ if (up) {
+ if (port-ep-umidi-chip-shutdown) {
+ /* gobble up remaining bytes to prevent wait in
+* snd_rawmidi_drain_output */
+ while (!snd_rawmidi_transmit_empty(substream))
+ snd_rawmidi_transmit_ack(substream, 1);
+ return;
+ }
tasklet_hi_schedule(port-ep-tasklet);
+ }
}
static int snd_usbmidi_input_open(snd_rawmidi_substream_t* substream)
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Re: [Alsa-devel] USB audio devices
Jaroslav Kysela wrote:
On Tue, 9 Mar 2004, Takashi Iwai wrote:
BTW, the USB audio is another headache. the current ALSA PCM model isn't
perfectly suitable for the devices like USB audio.
Unfortunately I don't see a better model.
Conceptually, USB devices have variable-sized periods. The question
is whether we actually want to allow this in the API. Probably not.
I have thought about forcing the period size to one ALSA frame (with
period_elapsed() calls at USB frame boundaries, like it's now). In
theory, applications should be able to cope with this.
I don't know much about USB 2.0,
Not too much differences for the driver. The main difference is that
there are now 8000 microframes per second. I have written a patch
(see below) and am about to test it.
but I guess that 1.x compatibility issues brings another overhead
to this extension which makes it also unuseable when 1.x devices
are connected to same USB port.
There are some nasty contortions when a 1.x device is connected
through a 2.0 hub, but otherwise there is no overhead for 1.x devices
because logically (and electrically AFAIK) either the 2.0 or the 1.x
host controller device is connected to a specific port.
Regards,
Clemens
--
Index: alsa-kernel/usb/usbaudio.c
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbaudio.c,v
retrieving revision 1.87
diff -u -r1.87 usbaudio.c
--- alsa-kernel/usb/usbaudio.c 8 Mar 2004 09:29:51 - 1.87
+++ alsa-kernel/usb/usbaudio.c 8 Mar 2004 09:48:09 -
@@ -104,6 +104,7 @@
*/
#define MAX_PACKS 10
+#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */
#define MAX_URBS 5 /* max. 20ms long packets */
#define SYNC_URBS 2 /* always two urbs for sync */
#define MIN_PACKS_URB 1 /* minimum 1 packet per urb */
@@ -165,6 +166,8 @@
unsigned int freqm; /* momentary sampling rate in USB format, i.e.
fs/1000 in Q10.14 */
unsigned int freqmax;/* maximum sampling rate, used for buffer management
*/
unsigned int phase; /* phase accumulator */
+ unsigned int phase_shift; /* fractional bits of phase */
+ unsigned int phase_mask;/* (1 phase_shift) - 1 */
unsigned int maxpacksize; /* max packet size in bytes */
unsigned int maxframesize; /* max packet size in frames */
unsigned int curpacksize; /* current packet size in bytes (for capture)
*/
@@ -250,7 +253,6 @@
cp[1] = subs-freqn 8;
cp[2] = subs-freqn 16;
}
- urb-interval = 1;
return 0;
}
@@ -301,7 +303,6 @@
spin_unlock_irqrestore(subs-lock, flags);
urb-transfer_buffer = ctx-buf;
urb-transfer_buffer_length = offs;
- urb-interval = 1;
#if 0 // for check
if (! urb-bandwidth) {
int bustime;
@@ -390,7 +391,6 @@
urb-iso_frame_desc[i].length = 3;
urb-iso_frame_desc[i].offset = offs;
}
- urb-interval = 1;
return 0;
}
@@ -464,8 +464,8 @@
if (subs-fill_max)
counts = subs-maxframesize; /* fixed */
else {
- subs-phase = (subs-phase 0x3fff) + subs-freqm;
- counts = subs-phase 14;
+ subs-phase = (subs-phase subs-phase_mask) + subs-freqm;
+ counts = subs-phase subs-phase_shift;
if (counts subs-maxframesize)
counts = subs-maxframesize;
}
@@ -515,7 +515,6 @@
spin_unlock_irqrestore(subs-lock, flags);
urb-transfer_buffer_length = offs * stride;
ctx-transfer = offs;
- urb-interval = 1;
return 0;
}
@@ -822,15 +821,22 @@
{
unsigned int maxsize, n, i;
int is_playback = subs-direction == SNDRV_PCM_STREAM_PLAYBACK;
- unsigned int npacks[MAX_URBS], total_packs;
+ unsigned int npacks[MAX_URBS], urb_packs, total_packs;
/* calculate the frequency in 10.14 format */
subs-freqn = subs-freqm = get_usb_rate(rate);
subs-freqmax = subs-freqn + (subs-freqn 2); /* max. allowed frequency */
subs-phase = 0;
+ if (subs-dev-speed == USB_SPEED_HIGH) {
+ subs-phase_shift = 11;
+ subs-phase_mask = 0x07ff;
+ } else {
+ subs-phase_shift = 14;
+ subs-phase_mask = 0x3fff;
+ }
/* calculate the max. size of packet */
- maxsize = ((subs-freqmax + 0x3fff) * (frame_bits 3)) 14;
+ maxsize = ((subs-freqmax + subs-phase_mask) * (frame_bits 3))
subs-phase_shift;
if (subs-maxpacksize maxsize subs-maxpacksize) {
//snd_printd(KERN_DEBUG maxsize %d is greater than defined size %d\n,
// maxsize, subs-maxpacksize);
@@ -842,9 +848,14 @@
else
Re: [Alsa-devel] USB audio devices
Jaroslav Kysela wrote: On Wed, 10 Mar 2004, Clemens Ladisch wrote: Conceptually, USB devices have variable-sized periods. The question is whether we actually want to allow this in the API. Probably not. What this does mean? I though that the period size is specified with time (1ms) for USB, isn't it? I think that we can describe this constraint with the refine algorithm. One USB frame is 1 ms, but the number of samples per USB frame varies if the sample rate isn't an integer multiple of 1000. E.g., at 44100 Hz, each USB frame has 44 samples, with one out of ten having 45. I don't know what happens when we constrain the period size to 1 ms, which would be exactly 44.1 frames. The period size in frames gets rounded to an integer, doesn't it? Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Full support for Creamware Noah and SFP
Willie Sippel wrote: But I'm not sure about, wether the NDA agrees in the basic alsa principles. What are the developers of alsa are thinking about that agreement? AFAIK drivers for ATI hardware are developed in a similar way (ATI gives specs under an NDA, the resulting driver is open source). Takashi, is that right? I told Frank about the work you did on Noah support, and he was _very_ impressed. Well, there isn't much reason to be impressed, as the Noah conforms to the USB audio specification, and the driver already implemented it. (The only reason I had to make a change to the driver was that the Noah and the driver, uh, didn't entirely agree on how to interpret a minor detail of the spec.) Tell him that NetBSD and OpenBSD have USB MIDI drivers, too. :-) Unlike the USB audio/MIDI parts, writing support for the vendor- specific parts of the Noah and for the SFP PCI cards won't be a matter of ten minutes, of course. It should work as follows: 1. Interested developers would sign the NDA. 2. Those developers would get the source and specs, and tell CW what parts need to become open to get the hardware working. Well, I would be interested in this. Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] USB audio devices
Takashi Iwai wrote: Clemens Ladisch wrote: I don't know much about USB 2.0, Not too much differences for the driver. The main difference is that there are now 8000 microframes per second. I have written a patch (see below) and am about to test it. is there any real usb2-audio device available? I have (remote) access to a machine with an Audigy 2 NX, which can do USB 2.0. And there is the Ediroal UA-1000, which I would really like to have access to. :-) Regards, Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] How do I redirect sound from 1st soundacrd to 2nd soundcard?
Bauke Jan Douma wrote: I have two soundcards in my system, a cmipci(card) and an intel8x0 (on board). I have the audio cable of my CD/ROM drive wired onto the cmipci. Attached to line-out of the cmipci is just my headphones. I also have a TV-card. An audio cable runs from the TV-card to the intel8x0. Attached to line-out of the intel8x0 is a speaker set. So when I play an audio CD (with cdctrl or xmcd or xmms) I usually listen to the audio on the headphones. This is intentional, because it suits me best. TV audio goes to the speaker set. However, occasionally I want to have the audio from a CD redirected to the speaker set. Right now, to accomplish this I do something like: $ cdda2wav -S 1 -t 1+12 -D /dev/sg0 -O wav - | \ aplay -D intel8x0 -f cd /dev/stdin These work without a problem, for the better part. But I was wondering if this cannot be done a bit more elegantly with some configuration in ~/.asoundrc, such that I can call aplay directly with that device as parameter to -D. aplay always reads its input from a file. IMHO, your pipe already is the most elegant way to provide a file for aplay. You can put it into a script. I would suggest to connect all input and output devices to one card, and to change the mixer settings appropriately. At least your onboard chip should be able to do this. HTH Clemens --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470alloc_id=3638op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] library problems
[EMAIL PROTECTED] wrote: I wanted to get the tempo, so I did this: snd_seq_queue_tempo_t *qt; snd_seq_queue_tempo_alloca(qt); snd_seq_get_queue_tempo(seq,id,qt); return qt-tempo;//line 91 and the compiler told me this: alsa.cpp:91: invalid use of undefined type `struct _snd_seq_queue_tempo' /usr/include/alsa/seq.h:382: forward declaration of `struct _snd_seq_queue_tempo' The real definition of snd_seq_queue_tempo_t is hidden deep inside the ALSA library. Applications are not allowed to access its fields directly to allow binary compatibility with future versions of ALSA. To get the tempo, call snd_seq_queue_tempo_get_tempo(qt). HTH Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Re: [PATCH] snd_pcm_hw_params_set_rate_near val pointer
Jack O'Quin wrote: Samuel S Chessman [EMAIL PROTECTED] writes: This is caused by snd_pcm_hw_params_set_rate_near parameter val needing to be a pointer, and alsa_driver passing in the frame rate instead. This occurs against current CVS of alsa. (2004-03-02) I think this is the wrong fix, Sam. Apparently, that parameter was changed in ALSA between 0.9 and 1.0. For compatibility with the maximum number of systems, the JACK alsa_driver is still compiled using ALSA_PCM_OLD_HW_PARAMS_API, which selects the older (non-pointer) interface. To elaborate on this: both ALSA 0.9 and 1.0 support both versions of these functions, and allow selecting between them by defining ALSA_PCM_OLD_HW_PARAMS_API or ALSA_PCM_NEW_HW_PARAMS_API. Without one of these symbols defined, the default would be the old API when compiling with 0.9 headers and the new one with 1.0 headers. All versions of the library are binary compatible with both old and new applications. So, maybe this is an ALSA bug in emulating the older interface? In theory, this should not happen. Sam, which version of ALSA are your ALSA headers/libraries from? Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Re: [alsa-cvslog] CVS: alsa-lib/src/pcm pcm.c,1.271,1.272
Abramo Bagnara wrote: Jaroslav Kysela ha scritto: Log Message: Fixed return comment for *near() functions + * \return 0 otherwise a negative error code if configuration space would become empty *near() functions semantic cannot return error by design. Taken for granted that configuration space on entry of function is not empty how it can become empty if I ask for *nearest* value? Obviously at least one configuration exists, then in the worst case such value is returned. It may be possible that other errors than an empty configuration space could occur. For example, future ALSA versions might want to fail if an hotplug device has been unpluggeed, or the configuration space could have been reduced due to some outside influence. I don't think we can guarantee that this function never ever fails. Maybe we should simply drop the if configuration space would become empty. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] fix for Creamware Noah
Hartmut Geissbauer wrote:
That _has_ helped. Now the device is fully functional.
- search class-specific endpoint descriptor after the sync ep, too
Index: alsa-kernel/usb/usbaudio.c
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbaudio.c,v
retrieving revision 1.81
diff -u -r1.81 usbaudio.c
--- alsa-kernel/usb/usbaudio.c 24 Feb 2004 08:44:33 - 1.81
+++ alsa-kernel/usb/usbaudio.c 27 Feb 2004 10:37:08 -
@@ -2324,6 +2324,9 @@
}
csep = snd_usb_find_desc(alts-endpoint[0].extra,
alts-endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+ /* Creamware Noah has this descriptor after the 2nd endpoint */
+ if (!csep altsd-bNumEndpoints = 2)
+ csep = snd_usb_find_desc(alts-endpoint[1].extra,
alts-endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
if (!csep || csep[0] 7 || csep[2] != EP_GENERAL) {
snd_printk(KERN_ERR %d:%u:%d : no or invalid class specific
endpoint descriptor\n,
dev-devnum, iface_no, altno);
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Re: [Alsa-devel] [partialy fixed] Creamware Noah, snd-usb-audio capable?
Hartmut Geissbauer wrote:
Clemens Ladisch wrote:
Hartmut Geissbauer wrote:
The only thing that's missing now are the two playback ports.
Please show the output of lsusb -v.
The descriptors look OK.
But only at first sight.
Please try this patch.
HTH
Clemens
--
Index: alsa-kernel/usb/usbaudio.c
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbaudio.c,v
retrieving revision 1.81
diff -u -r1.81 usbaudio.c
--- alsa-kernel/usb/usbaudio.c 24 Feb 2004 08:44:33 - 1.81
+++ alsa-kernel/usb/usbaudio.c 26 Feb 2004 10:21:55 -
@@ -2324,6 +2324,9 @@
}
csep = snd_usb_find_desc(alts-endpoint[0].extra,
alts-endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+ /* Creamware Noah has this descriptor after the 2nd endpoint */
+ if (!csep altsd-bNumEndpoints = 2)
+ csep = snd_usb_find_desc(alts-endpoint[1].extra,
alts-endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
if (!csep || csep[0] 7 || csep[2] != EP_GENERAL) {
snd_printk(KERN_ERR %d:%u:%d : no or invalid class specific
endpoint descriptor\n,
dev-devnum, iface_no, altno);
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Re: [Alsa-devel] period_size, periods, count parameters
Mathieu Rondonneau wrote: I don't understand why count is 1024 ? why count is not equal to runtime-period_size (i.e. 16*1024)? If your frame size is 4, this is the size of a memory page. I guess ALSA wants to copy the data one page at a time, and this is required if the buffer isn't contiguous. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] [PATCH] add aplaymidi arecordmidi utils
Will wrote: Clemens Ladisch [EMAIL PROTECTED] wrote: If you get correct output when you run amidi -p virtual -d and aconnect your keyboard to that port, then it may be a bug in arecordmidi. I'll test when I'm at home. This is what I get: $ pmidi -l Port Client name Port name 64:0 Rawmidi 0 - Sound Blaster 16 MSound Blaster 16 MPU-401 You didn't apply all my patches, or recompile; this name would be different. Did you change asequencer.h in both the kernel and alsa-lib? playing various notes on the MIDI keyboard gives correct MIDI output in ASCII: 90 29 4A ... On my machine, arecordmidi works just fine. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] fix amixer load error messages
fix error messages for snd_mixer_load
BTW: yesterday's change to alsa-lib's asequencer.h hasn't been
committed yet
Index: alsa-utils/amixer/amixer.c
===
RCS file: /cvsroot/alsa/alsa-utils/amixer/amixer.c,v
retrieving revision 1.79
diff -u -r1.79 amixer.c
--- alsa-utils/amixer/amixer.c 21 Feb 2004 18:44:18 - 1.79
+++ alsa-utils/amixer/amixer.c 24 Feb 2004 08:56:13 -
@@ -774,7 +774,7 @@
}
err = snd_mixer_load(handle);
if (err 0) {
- error(Mixer load error: %s, card, snd_strerror(err));
+ error(Mixer %s load error: %s, card, snd_strerror(err));
snd_mixer_close(handle);
return err;
}
@@ -1145,7 +1145,7 @@
}
err = snd_mixer_load(handle);
if (err 0) {
- error(Mixer load error: %s, card, snd_strerror(err));
+ error(Mixer %s load error: %s, card, snd_strerror(err));
snd_mixer_close(handle);
return err;
}
@@ -1427,7 +1427,7 @@
snd_mixer_set_callback(handle, mixer_event);
err = snd_mixer_load(handle);
if (err 0) {
- error(Mixer load error: %s, card, snd_strerror(err));
+ error(Mixer %s load error: %s, card, snd_strerror(err));
snd_mixer_close(handle);
return err;
}
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Re: [Alsa-devel] [PATCH] add aplaymidi arecordmidi utils
Will wrote: How easy would it be to make aplaymidi be able to adjust the playback speed like playmidi? (playmidi -t 0.5 music.mid plays at twice the default speed) Very easy. How easy would it be to make aplaymidi be able to move forwards and backwards during playback, e.g. by pressing f or b, like Takashi Iwai's drvmidi? Not so easy; even more so if you want to have correct controller values. Adjusting settings changed by sysexs would be almost impossible. However, these tools are meant as basic command-line utilities (and examples). I think your suggestions would fit better into some GUI player. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] [PATCH] add aplaymidi arecordmidi utils
Will wrote: Clemens Ladisch [EMAIL PROTECTED] wrote: I'm getting empty MIDI files containing only the MIDI header. Are you using the latest CVS version of ALSA? Some of the bugfixes I sent yesterday are needed by arecordmidi. No, but I applied your patches to alsa-lib-1.0.3rc1 and Linux 2.6.3 in include/sound/asequencer, sound/* etc. Is that enough to get a working ALSA sequencer? If you recompiled both alsa-driver and alsa-lib packages, yes. Perhaps I should test more; the latest tests were made only against data recorded through the snd-seq-dummy client (this is where the unpatched sequencer would fail). If you get correct output when you run amidi -p virtual -d and aconnect your keyboard to that port, then it may be a bug in arecordmidi. I'll test when I'm at home. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Compile errors with latest alsa releases
snookertb wrote: SND_CTL_READONLY does not appear to be declared anywhere. It's defined in alsa-lib/include/control.h. You have to install the alsa-lib package before compiling alsa-utils and alsa-tools. Are you part of the alsa development team? SourceForge says no, the bugtracker says yes. :-) Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] period_size, periods, count parameters
Mathieu Rondonneau wrote: .period_bytes_min = 1092, .period_bytes_max = 32768, .periods_min = 8, .periods_max = 512, When i start to send a file, I Don't know why: runtime-period_size = 8192 runtime-periods: 8 and the ¨count¨ parameter in the copy operator is 1024 These values fit into the constraints defined above. I tryed to change .period_bytes_min = 8192, .periods_min= 1, but the ¨count¨ parameter, the ¨periode_size¨ and ¨periods¨ remain the same. The same for these values. Is it possible to change the ¨count¨ parameter in the copy operator? Apparently, your copy operator is to copy one period. Probably you have to force another period size. What's wrong with these parameters? Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id56alloc_id438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] change snd-seq-midi client names
This patch reverses the order of the Rawmidi x and rawmidi name parts of client names to enable selecting clients by a unique prefix (as snd_seq_parse_address does). Index: alsa-kernel/core/seq/seq_midi.c === RCS file: /cvsroot/alsa/alsa-kernel/core/seq/seq_midi.c,v retrieving revision 1.15 diff -u -r1.15 seq_midi.c --- alsa-kernel/core/seq/seq_midi.c 7 Jan 2004 14:34:22 - 1.15 +++ alsa-kernel/core/seq/seq_midi.c 23 Feb 2004 08:27:41 - @@ -285,7 +285,7 @@ cinfo.client = client-seq_client; cinfo.type = KERNEL_CLIENT; name = rmidi-name[0] ? (const char *)rmidi-name : External MIDI; - snprintf(cinfo.name, sizeof(cinfo.name), Rawmidi %d - %s, card-number, name); + snprintf(cinfo.name, sizeof(cinfo.name), %s - Rawmidi %d, name, card-number); return snd_seq_kernel_client_ctl(client-seq_client, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, cinfo); } --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] add USB MIDI port names
- add device-specific port names
- begin numbering ports at 1
Index: alsa-kernel/usb/usbmidi.c
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbmidi.c,v
retrieving revision 1.22
diff -u -r1.22 usbmidi.c
--- alsa-kernel/usb/usbmidi.c 26 Jan 2004 14:19:24 - 1.22
+++ alsa-kernel/usb/usbmidi.c 23 Feb 2004 08:27:42 -
@@ -1,7 +1,7 @@
/*
* usbmidi.c - ALSA USB MIDI driver
*
- * Copyright (c) 2002 Clemens Ladisch
+ * Copyright (c) 2002-2004 Clemens Ladisch
* All rights reserved.
*
* Based on the OSS usb-midi driver by NAGANO Daisuke,
@@ -727,18 +727,92 @@
return NULL;
}
+/*
+ * This list specifies names for ports that do not fit into the standard
+ * (product) MIDI (n) schema because they aren't external MIDI ports,
+ * such as internal control or synthesizer ports.
+ */
+static struct {
+ __u16 vendor;
+ __u16 product;
+ int port;
+ const char *name_format;
+} snd_usbmidi_port_names[] = {
+ /* Roland UA-100 */
+ {0x0582, 0x, 2, %s Control},
+ /* Roland SC-8850 */
+ {0x0582, 0x0003, 0, %s Part A},
+ {0x0582, 0x0003, 1, %s Part B},
+ {0x0582, 0x0003, 2, %s Part C},
+ {0x0582, 0x0003, 3, %s Part D},
+ {0x0582, 0x0003, 4, %s MIDI 1},
+ {0x0582, 0x0003, 5, %s MIDI 2},
+ /* Roland U-8 */
+ {0x0582, 0x0004, 1, %s Control},
+ /* Roland SC-8820 */
+ {0x0582, 0x0007, 0, %s Part A},
+ {0x0582, 0x0007, 1, %s Part B},
+ {0x0582, 0x0007, 2, %s MIDI},
+ /* Roland SK-500 */
+ {0x0582, 0x000b, 0, %s Part A},
+ {0x0582, 0x000b, 1, %s Part B},
+ {0x0582, 0x000b, 2, %s MIDI},
+ /* Roland SC-D70 */
+ {0x0582, 0x000c, 0, %s Part A},
+ {0x0582, 0x000c, 1, %s Part B},
+ {0x0582, 0x000c, 2, %s MIDI},
+ /* Edirol UM-880 */
+ {0x0582, 0x0014, 8, %s Control},
+ /* Edirol SD-90 */
+ {0x0582, 0x0016, 0, %s Part A},
+ {0x0582, 0x0016, 1, %s Part B},
+ {0x0582, 0x0016, 2, %s MIDI 1},
+ {0x0582, 0x0016, 3, %s MIDI 2},
+ /* Edirol UM-550 */
+ {0x0582, 0x0023, 5, %s Control},
+ /* Edirol SD-20 */
+ {0x0582, 0x0027, 0, %s Part A},
+ {0x0582, 0x0027, 1, %s Part B},
+ {0x0582, 0x0027, 2, %s MIDI},
+ /* Edirol SD-80 */
+ {0x0582, 0x0029, 0, %s Part A},
+ {0x0582, 0x0029, 1, %s Part B},
+ {0x0582, 0x0029, 2, %s MIDI 1},
+ {0x0582, 0x0029, 3, %s MIDI 2},
+ /* M-Audio MidiSport 8x8 */
+ {0x0763, 0x1031, 8, %s Control},
+ {0x0763, 0x1033, 8, %s Control},
+};
+
static void snd_usbmidi_init_substream(snd_usb_midi_t* umidi,
int stream, int number,
snd_rawmidi_substream_t** rsubstream)
{
+ int i;
+ __u16 vendor, product;
+ const char *name_format;
+
snd_rawmidi_substream_t* substream = snd_usbmidi_find_substream(umidi, stream,
number);
if (!substream) {
snd_printd(KERN_ERR substream %d:%d not found\n, stream, number);
return;
}
+
/* TODO: read port name from jack descriptor */
+ name_format = %s MIDI %d;
+ vendor = umidi-chip-dev-descriptor.idVendor;
+ product = umidi-chip-dev-descriptor.idProduct;
+ for (i = 0; i ARRAY_SIZE(snd_usbmidi_port_names); ++i) {
+ if (snd_usbmidi_port_names[i].vendor == vendor
+ snd_usbmidi_port_names[i].product == product
+ snd_usbmidi_port_names[i].port == number) {
+ name_format = snd_usbmidi_port_names[i].name_format;
+ break;
+ }
+ }
snprintf(substream-name, sizeof(substream-name),
-%s Port %d, umidi-chip-card-shortname, number);
+name_format, umidi-chip-card-shortname, number + 1);
+
*rsubstream = substream;
}
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[Alsa-devel] [PATCH] fix KERNEL_QUOTE event handling
Timestamping (if enabled on a subscription or a port) is not applied
to the quoted event but to the quoting event. This patch adds a
function to copy only selected fields into the event to be delivered.
- fix KERNEL_QUOTE event timestamping
- fix typo in port_broadcast_event
Index: alsa-kernel/core/seq/seq_clientmgr.c
===
RCS file: /cvsroot/alsa/alsa-kernel/core/seq/seq_clientmgr.c,v
retrieving revision 1.27
diff -u -r1.27 seq_clientmgr.c
--- alsa-kernel/core/seq/seq_clientmgr.c5 Feb 2004 15:38:42 - 1.27
+++ alsa-kernel/core/seq/seq_clientmgr.c23 Feb 2004 08:27:41 -
@@ -547,6 +547,36 @@
/*
+ * expand a quoted event.
+ */
+static int expand_quoted_event(snd_seq_event_t *event)
+{
+ snd_seq_event_t *quoted;
+
+ quoted = event-data.quote.event;
+ if (quoted == NULL) {
+ snd_printd(seq: quoted event is NULL\n);
+ return -EINVAL;
+ }
+
+ event-type = quoted-type;
+ event-tag = quoted-tag;
+ event-source = quoted-source;
+ /* don't use quoted destination */
+ event-data = quoted-data;
+ /* use quoted timestamp only if subscription/port didn't update it */
+ if (event-queue == SNDRV_SEQ_QUEUE_DIRECT) {
+ event-flags = quoted-flags;
+ event-queue = quoted-queue;
+ event-time = quoted-time;
+ } else {
+ event-flags = (event-flags SNDRV_SEQ_TIME_STAMP_MASK)
+ | (quoted-flags ~SNDRV_SEQ_TIME_STAMP_MASK);
+ }
+ return 0;
+}
+
+/*
* deliver an event to the specified destination.
* if filter is non-zero, client filter bitmap is tested.
*
@@ -581,12 +611,9 @@
update_timestamp_of_queue(event, dest_port-time_queue,
dest_port-time_real);
- /* expand the quoted event */
if (event-type == SNDRV_SEQ_EVENT_KERNEL_QUOTE) {
quoted = 1;
- event = event-data.quote.event;
- if (event == NULL) {
- snd_printd(seq: quoted event is NULL\n);
+ if (expand_quoted_event(event) 0) {
result = 0; /* do not send bounce error */
goto __skip;
}
@@ -694,8 +721,8 @@
if (dest_client == NULL)
return 0; /* no matching destination */
- read_lock(client-ports_lock);
- list_for_each(p, client-ports_list_head) {
+ read_lock(dest_client-ports_lock);
+ list_for_each(p, dest_client-ports_list_head) {
client_port_t *port = list_entry(p, client_port_t, list);
event-dest.port = port-addr.port;
/* pass NULL as source client to avoid error bounce */
@@ -706,7 +733,7 @@
break;
num_ev++;
}
- read_unlock(client-ports_lock);
+ read_unlock(dest_client-ports_lock);
snd_seq_client_unlock(dest_client);
event-dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */
return (err 0) ? err : num_ev;
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[Alsa-devel] [PATCH] fix MPU-401 port names
remove unneeded technical information from port names Index: alsa-kernel/drivers/mpu401/mpu401_uart.c === RCS file: /cvsroot/alsa/alsa-kernel/drivers/mpu401/mpu401_uart.c,v retrieving revision 1.29 diff -u -r1.29 mpu401_uart.c --- alsa-kernel/drivers/mpu401/mpu401_uart.c13 Jan 2004 17:11:25 - 1.29 +++ alsa-kernel/drivers/mpu401/mpu401_uart.c23 Feb 2004 08:27:41 - @@ -528,9 +528,9 @@ mpu-irq = irq; mpu-irq_flags = irq_flags; if (card-shortname[0]) - snprintf(rmidi-name, sizeof(rmidi-name), %s MPU-401, card-shortname); + snprintf(rmidi-name, sizeof(rmidi-name), %s MIDI, card-shortname); else - sprintf(rmidi-name, MPU-401 (UART) %d-%d, card-number, device); + sprintf(rmidi-name, MPU-401 MIDI %d-%d, card-number, device); snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, snd_mpu401_uart_output); snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, snd_mpu401_uart_input); rmidi-info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] fix typo in port flags
- fix typo in port flags
- add GM2 capability bit
Index: alsa-kernel/include/asequencer.h
===
RCS file: /cvsroot/alsa/alsa-kernel/include/asequencer.h,v
retrieving revision 1.11
diff -u -r1.11 asequencer.h
--- alsa-kernel/include/asequencer.h7 Jan 2004 14:34:23 - 1.11
+++ alsa-kernel/include/asequencer.h23 Feb 2004 08:27:42 -
@@ -594,6 +594,7 @@
#define SNDRV_SEQ_PORT_TYPE_MIDI_GS(13) /* GS compatible device */
#define SNDRV_SEQ_PORT_TYPE_MIDI_XG(14) /* XG compatible device */
#define SNDRV_SEQ_PORT_TYPE_MIDI_MT32 (15) /* MT-32 compatible device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_GM2 (16) /* General MIDI 2 compatible device */
/* other standards...*/
#define SNDRV_SEQ_PORT_TYPE_SYNTH (110) /* Synth device (no MIDI compatible -
direct wavetable) */
@@ -605,7 +606,7 @@
/* misc. conditioning flags */
#define SNDRV_SEQ_PORT_FLG_GIVEN_PORT (10)
#define SNDRV_SEQ_PORT_FLG_TIMESTAMP (11)
-#define SNDRV_SEQ_PORT_FLG_TIME_REAL (11)
+#define SNDRV_SEQ_PORT_FLG_TIME_REAL (12)
struct sndrv_seq_port_info {
struct sndrv_seq_addr addr; /* client/port numbers */
Index: alsa-lib/include/sound/asequencer.h
===
RCS file: /cvsroot/alsa/alsa-lib/include/sound/asequencer.h,v
retrieving revision 1.3
diff -u -r1.3 asequencer.h
--- alsa-lib/include/sound/asequencer.h 7 Jan 2004 14:36:40 - 1.3
+++ alsa-lib/include/sound/asequencer.h 23 Feb 2004 08:27:42 -
@@ -594,6 +594,7 @@
#define SNDRV_SEQ_PORT_TYPE_MIDI_GS(13) /* GS compatible device */
#define SNDRV_SEQ_PORT_TYPE_MIDI_XG(14) /* XG compatible device */
#define SNDRV_SEQ_PORT_TYPE_MIDI_MT32 (15) /* MT-32 compatible device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_GM2 (16) /* General MIDI 2 compatible device */
/* other standards...*/
#define SNDRV_SEQ_PORT_TYPE_SYNTH (110) /* Synth device (no MIDI compatible -
direct wavetable) */
@@ -605,7 +606,7 @@
/* misc. conditioning flags */
#define SNDRV_SEQ_PORT_FLG_GIVEN_PORT (10)
#define SNDRV_SEQ_PORT_FLG_TIMESTAMP (11)
-#define SNDRV_SEQ_PORT_FLG_TIME_REAL (11)
+#define SNDRV_SEQ_PORT_FLG_TIME_REAL (12)
struct sndrv_seq_port_info {
struct sndrv_seq_addr addr; /* client/port numbers */
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[Alsa-devel] [PATCH] add aplaymidi arecordmidi utils
See man pages for details. add aplaymidi arecordmidi utilities Index: alsa-utils/configure.in === RCS file: /cvsroot/alsa/alsa-utils/configure.in,v retrieving revision 1.76 diff -u -r1.76 configure.in --- alsa-utils/configure.in 26 Jan 2004 19:30:45 - 1.76 +++ alsa-utils/configure.in 23 Feb 2004 08:27:54 - @@ -69,4 +69,5 @@ AC_OUTPUT(Makefile alsactl/Makefile alsamixer/Makefile amidi/Makefile amixer/Makefile \ alsaconf/alsaconf alsaconf/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ - utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile seq/aseqnet/Makefile) + utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ + seq/aplaymidi/Makefile seq/aseqnet/Makefile) Index: alsa-utils/amidi/amidi.1 === RCS file: /cvsroot/alsa/alsa-utils/amidi/amidi.1,v retrieving revision 1.2 diff -u -r1.2 amidi.1 --- alsa-utils/amidi/amidi.119 Jan 2004 18:39:12 - 1.2 +++ alsa-utils/amidi/amidi.123 Feb 2004 08:27:54 - @@ -1,4 +1,4 @@ -.TH AMIDI 1 18 Jan 2004 +.TH AMIDI 1 22 Feb 2004 .SH NAME amidi \- read from and write to ALSA RawMIDI ports @@ -16,7 +16,12 @@ .B amidi handles only files containing raw MIDI commands, without timing information. -Standard MIDI (.mid) files are not supported. +.B amidi +does not support Standard MIDI (.mid) files, but +.B aplaymidi(1) +and +.B arecordmidi(1) +do. .SH INVOKING .B amidi @@ -68,17 +73,15 @@ .I -s, --send=filename Sends the contents of the specified file to the MIDI port. The file must contain raw MIDI commands (e.g. a .syx file); -you can -.I not -use a Standard MIDI (.mid) file for this. +for Standard MIDI (.mid) files, use +.B aplaymidi(1). .TP .I -r, --receive=filename Writes data received from the MIDI port into the specified file. The file will contain raw MIDI commands (such as in a .syx file); -this will -.I not -create a Standard MIDI (.mid) file. +to record a Standard MIDI (.mid) file, use +.B arecordmidi(1). .B amidi will filter out any Active Sensing bytes (FEh), unless the @@ -155,6 +158,11 @@ The .I --list-devices option pretends that output and input ports are the same. + +.SH SEE ALSO +aplaymidi(1) +.br +arecordmidi(1) .SH AUTHOR Clemens Ladisch [EMAIL PROTECTED] Index: alsa-utils/seq/Makefile.am === RCS file: /cvsroot/alsa/alsa-utils/seq/Makefile.am,v retrieving revision 1.1 diff -u -r1.1 Makefile.am --- alsa-utils/seq/Makefile.am 4 Jan 2000 12:39:00 - 1.1 +++ alsa-utils/seq/Makefile.am 23 Feb 2004 08:27:54 - @@ -1 +1 @@ -SUBDIRS=aconnect aseqnet +SUBDIRS=aconnect aplaymidi aseqnet diff -urN ../../oalsa/alsa/alsa-utils/seq/aplaymidi/Makefile.am alsa-utils/seq/aplaymidi/Makefile.am --- ../../oalsa/alsa/alsa-utils/seq/aplaymidi/Makefile.am 1970-01-01 01:00:00.0 +0100 +++ alsa-utils/seq/aplaymidi/Makefile.am2004-02-23 09:38:11.051239000 +0100 @@ -0,0 +1,5 @@ +INCLUDES = -I$(top_srcdir)/include +EXTRA_DIST = aplaymidi.1 arecordmidi.1 + +bin_PROGRAMS = aplaymidi arecordmidi +man_MANS = aplaymidi.1 arecordmidi.1 diff -urN ../../oalsa/alsa/alsa-utils/seq/aplaymidi/aplaymidi.1 alsa-utils/seq/aplaymidi/aplaymidi.1 --- ../../oalsa/alsa/alsa-utils/seq/aplaymidi/aplaymidi.1 1970-01-01 01:00:00.0 +0100 +++ alsa-utils/seq/aplaymidi/aplaymidi.12004-02-23 09:38:11.061243000 +0100 @@ -0,0 +1,55 @@ +.TH APLAYMIDI 1 15 Feb 2004 + +.SH NAME +aplaymidi \- play Standard MIDI Files + +.SH SYNOPSIS +.B aplaymidi +-p client:port[,...] [-d delay] midifile ... + +.SH DESCRIPTION +.B aplaymidi +is a command-line utility that plays the specified MIDI file(s) to one +or more ALSA sequencer ports. + +.SH OPTIONS + +.TP +.I -h, --help +Prints a list of options. + +.TP +.I -V, --version +Prints the current version. + +.TP +.I -l, --list +Prints a list of possible output ports. + +.TP +.I -p, --port=client:port,... +Sets the sequencer port(s) to which the events in the MIDI file(s) are +sent. + +A client can be specified by its number, its name, or a prefix of its +name. A port is specified by its number; for port 0 of a client, the +:0 part of the port specification can be omitted. + +For compatibility with +.B pmidi(1), +the port specification is taken from the +.I ALSA_OUTPUT_PORTS +environment variable if none is given on the command line. + +.TP +.I -d, --delay=seconds +Specifies how long to wait after the end of each MIDI file, +to allow the last notes to die away. + +.SH SEE ALSO +pmidi(1) +.br +playmidi(1) + +.SH AUTHOR +Clemens Ladisch [EMAIL PROTECTED] diff -urN ../../oalsa/alsa/alsa-utils/seq/aplaymidi/aplaymidi.c alsa-utils/seq/aplaymidi/aplaymidi.c --- ../../oalsa/alsa/alsa-utils/seq/aplaymidi/aplaymidi.c 1970-01-01 01:00:00.0 +0100 +++ alsa-utils/seq/aplaymidi
Re: [Alsa-devel] Compile errors with latest alsa releases
snookertb wrote: I am attempting to update the Sun JDS distro from alsa version 0.9.0 to a later version which supports newer Intel sound chips. I get the following errors in these two compiles. /home/tarpon/downloads/alsa-driver-1.0.2c/include/adriver.h:295: redefinition of `usb_make_path' This is a function which doesn't exist in older 2.4 kernels, so ALSA defines its own version. It seems Sun backported some USB stuff from a newer kernel. Try to uncomment this function in adriver.h. HTH Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] emagic mt4
ian esten wrote: thanks clemens. the mt4 now shows up in /proc/asound/cards. unfortunately though it doesnt seem to be working. i sent it some midi from seq24, and got no blinking lights on the front of the unit. do you think it will be possible to get the mt4 working with alsa? It's possible, but not easy. We have to find out what protocol the device uses. You could install something like usbsnoop on a Windows computer and send/receive a few MIDI bytes, then send me the log file, then I could try to find out what the stuff means. If you do this, please send no more than a few MIDI commands, and don't connect any other USB devices, because the log file can become very big. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] emagic mt4
ian esten wrote:
On Thu, 2004-02-19 at 03:28, Clemens Ladisch wrote:
Please post the output of lsusb -v.
Bus 001 Device 040: ID 086a:0003 Emagic Soft-und Hardware GmbH
.
.
.
These descriptors look very much like those of earlier Edirol devices.
It may be possible that the MT4 is similar enough to speak the same
protocol.
Please apply the patch below and recompile ALSA. The device and its
ports will show up in ALSA, but I'm not sure if sending/receiving MIDI
data will work.
HTH
Clemens
--
Index: alsa-kernel/usb/usbquirks.h
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbquirks.h,v
retrieving revision 1.28
diff -u -r1.28 usbquirks.h
--- alsa-kernel/usb/usbquirks.h 18 Feb 2004 16:24:55 - 1.28
+++ alsa-kernel/usb/usbquirks.h 20 Feb 2004 11:15:09 -
@@ -714,4 +714,19 @@
},
+/* Emagic devices */
+{
+ USB_DEVICE(0x086a, 0x0003),
+ .driver_info = (unsigned long) (const snd_usb_audio_quirk_t) {
+ .vendor_name = Emagic,
+ .product_name = MT4,
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x000f,
+ .in_cables = 0x0003
+ }
+ }
+},
+
{
---
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Re: [Alsa-devel] [partialy fixed] Creamware Noah, snd-usb-audio capable?
Hartmut Geissbauer wrote: Clemens Ladisch wrote: Might the patch(the code in it) for usbaudio.c you posted before be helpfull for the program? No. Please remove the patch. Done, except of the entries in usbquirks.h. Is it still needed? No. The quirk entry causes the driver to be loaded even for the first configuration, which isn't needed anymore. Now by playing around with the settings in the build-in menu of the synthesizer, I discovered an entry named USBCom, that isn't described in the (so far well done) manual. You can switch it between Windows and Mac/Linux. I rememberd that I updated the firmware. (Now: 1.4.23, if anyone else is interested) So this menu entry might belong to this update. OK, with this setting, it is _not_ necessary to switch the configuration with your small but excellent tool to 2. There's now only _one_ configuration available, which represents the audio and midi devices. It's nice to see a device with built-in Linux support. :-) Please make an entry on http://qbik.ch/usb/devices/ and describe to change the USBCom entry. The only thing that's missing now are the two playback ports. Please show the output of lsusb -v. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] emagic mt4
ian esten wrote: i have an emagic mt4 i'm trying to use with alsa. Currently, nobody knows how to communicate with this device. Please post the output of lsusb -v. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] crashes usb-audio yamaha rp-u200
jaap wrote: When I start jack with jackstart -d alsa -P hw:0,0 -C hw:0,1, jack seems to start ,but then the system freezes with kernel panic.Sometimes jack is running but then the system crashes as soon as I try to do something with audio alsa-driver 0.9.8 This is quite old. Please try the latest version. Adhoc cc95e706 [uhci]rh_int_timer_do+76/1e0 Uh, this isn't ALSA, it's in the USB driver. :-) But this could be the async unlinking bug where ALSA was responsible (this has been fixed recently). HTH Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] ad1888 as in ASUS P4R800-VM motherboards
Peter Antypas wrote: I 've had no luck in using the cvs version with the ad1888 update. In fact, I can't even load snd-intel8x0. I thought the intel8x0 module was supposed to work here? An AC'97 system consists of two components: an AC'97 codec (in your case, the AD1888), and an AC'97 controller (usually part of the southbridge). Your codec is supported, but your controller isn't. cat /proc/pci reveals a PCI id of 1002:4341 with I/O at 0xfbe0 to 0xfbe000ff. It reports the device is (ATI Technologies) (rev 0) grepping through the CVS source (checked out today) for 1002 or 4341 reveals nothing. I added the PCI ids to snd-intel8x0 and it fails to grab the specified I/O ports. What could be the problem here? The problem is that ATI's AC'97 controller isn't compatible with Intel's, so snd-intel8x0 doesn't work with it. And it's completely undocumented, so we cannot write a driver for it. Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Creamware Noah, snd-usb-audio capable?
Hartmut Geissbauer wrote: Clemens Ladisch wrote: The problem is that the kernel automatically selects the first configuration. Please try the patch below. drivers/usb/core/usb.c: usb_hotplug usb 1-2.4: configuration #1 chosen from 2 choices usb 1-2.4: registering 1-2.4:1.0 (config #1, interface 0) drivers/usb/core/usb.c: usb_hotplug snd-usb-audio 1-2.4:1.0: usb_probe_interface snd-usb-audio 1-2.4:1.0: usb_probe_interface - got id The device is not appearing in /proc/asound. Does the new configuration (the audio interfaces) show up in /proc/bus/usb/devices? What happens when you re-load snd-usb-audio after that? Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Creamware Noah, snd-usb-audio capable?
Frank Barknecht wrote: Clemens Ladisch hat gesagt: // Clemens Ladisch wrote: it should be possible to use the MIDI_STANDARD_INTERFACE quirk (see below). I'd love to test it, but I had to send back the device already. :( Could you test it? I would test it if I actually had the device. Well, it's in CVS now, so let's pretend it works and see if somebody complains. ;-) Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Creamware Noah, snd-usb-audio capable?
Hartmut Geissbauer wrote:
I own a piece of hardware called Noah EX. It's a tactive
instrument modeller.
My goal is to get this device working with the alsa usb driver,
because the output of lsusb -v leads me to the hope, that it has a
standard usb audio interface.
...
Interface Descriptor:
bInterfaceClass 1 Audio
bInterfaceSubClass 1 Control Device
...
bInterfaceSubClass 2 Streaming (= PCM)
...
bInterfaceSubClass 3 Non Streaming (= MIDI)
Well, this device indeed has standard descriptors.
What I discoverd yet, is that the device is offering two
configurations. And the second is the audio and midi related one.
The first seems to be the configuration channel.
Feb 5 00:28:33 chief kernel: usb 1-2.4: configuration #1 chosen from 2 choices
The problem is that the kernel automatically selects the first
configuration. Please try the patch below.
HTH
Clemens
--
Index: alsa-kernel/usb/usbquirks.h
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbquirks.h,v
retrieving revision 1.27
diff -u -r1.27 usbquirks.h
--- alsa-kernel/usb/usbquirks.h 11 Feb 2004 12:15:55 - 1.27
+++ alsa-kernel/usb/usbquirks.h 12 Feb 2004 10:56:21 -
@@ -698,4 +698,13 @@
}
},
+{
+ USB_DEVICE_VENDOR_SPEC(0x1107, 0x0102),
+ .driver_info = (unsigned long) (const snd_usb_audio_quirk_t) {
+ .vendor_name = CreamWare,
+ .product_name = Noah,
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+
#undef USB_DEVICE_VENDOR_SPEC
Index: alsa-kernel/usb/usbaudio.c
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbaudio.c,v
retrieving revision 1.79
diff -u -r1.79 usbaudio.c
--- alsa-kernel/usb/usbaudio.c 9 Feb 2004 16:25:38 - 1.79
+++ alsa-kernel/usb/usbaudio.c 12 Feb 2004 10:56:21 -
@@ -2820,6 +2820,14 @@
goto __err_val;
config = dev-actconfig;
}
+ /* Creamware Noah needs to be set to the second configuration */
+ if (dev-descriptor.idVendor == 0x1107 dev-descriptor.idProduct == 0x0102
+ ifnum == 0) {
+ if (get_cfg_desc(dev-actconfig)-bConfigurationValue == 1)
+ usb_set_configuration(dev, 2);
+ /* but don't actually claim interface 0 */
+ goto __err_val;
+ }
/*
* found a config. now register to ALSA
---
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[Alsa-devel] [PATCH] UA-3FX support
Frank Barknecht wrote:
Now I discovered a teenytiny switch on the back if the UA-3 FX that
switches the Advanced Driver features of the device on or off.
Setting this to OFF, I now got ALSA to recognize the device as a
simple 44100 Hz device. No higher samplerates, though.
Probably a quirk is needed?
The descriptors in advanced mode look like this:
unknown descriptor type: 0b 24 02 01 02 03 18 01 44 ac 00
which is 24 bits at 44.1 kHz. Probably there is another switch for
the sample rate. :-)
The device has another interface (3) without any descriptors. You
wouldn't know, just by chance, if the Windows driver offers some MIDI
port, or some device-specific control application for the effects?
Regards,
Clemens
--
add UA-3FX support
Index: alsa-kernel/usb/usbquirks.h
===
RCS file: /cvsroot/alsa/alsa-kernel/usb/usbquirks.h,v
retrieving revision 1.27
diff -u -r1.27 usbquirks.h
--- alsa-kernel/usb/usbquirks.h 11 Feb 2004 12:15:55 - 1.27
+++ alsa-kernel/usb/usbquirks.h 12 Feb 2004 11:19:01 -
@@ -502,6 +502,32 @@
}
},
{
+ /*
+* This quirk is for the Advanced Driver mode. If off, the UA-3FX
+* is standard compliant, but has only 16-bit PCM.
+*/
+ USB_DEVICE(0x0582, 0x0050),
+ .driver_info = (unsigned long) (const snd_usb_audio_quirk_t) {
+ .vendor_name = EDIROL,
+ .product_name = UA-3FX,
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
USB_DEVICE(0x0582, 0x0052),
.driver_info = (unsigned long) (const snd_usb_audio_quirk_t) {
.vendor_name = EDIROL,
---
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Re: [Alsa-devel] [Fwd: Audigy2 NX (USB 2.0) on ALSA 1.0.2]
John W. Cocula wrote: I have an Audigy2 NX working under ALSA 1.0.2 (the latest 1.0.2 :) ... 2. I downloaded the latest firmware under Windows that supposedly permits USB 2.0 operation, but lsusb -v stills shows it as a 1.1 device. Bus 002 Device 004: ID 041e:3020 Creative Technology, Ltd ... bcdUSB 1.10 Yes, this is USB 1.1, the device says so itself. Am I right in assuming that this has absolutely nothing to do with ALSA? Yes. Does anyone have any pointers on making this a true USB 2.0 connection? Probably the device is now compatible with USB 2.0, i.e., the firmware update prevents the device from blowing up when connected to a 2.0-capable computer, or something like that. Or the device is 1.1 by default and switches to 2.0 only when the driver sends it some special command. (The Extigy needs a similar hack to enable some advanced features.) Could you check if the Windows device manager shows it connected to the 2.0 controller (EHCI) or a 1.x controller (UHCI or OHCI)? Regards, Clemens --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356alloc_id=3438op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
