October 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] Fix for assertion in pulse plugin when calling hw_params on a prepared stream
by Sean McNamara 29 Oct '07

29 Oct '07

Hello, I have opened a bug and recently submitted a patch for it on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3470 All the details are there. But here is a 5-minute summary for you: --The Problem-- (1) All plugins implementing the PCM interface, that I've tested so far, allow calling snd_*_hw_params on a stream when snd_*_state returns SND_PCM_STATE_PREPARED. (2) Applications have used this fact to their advantage, and sometimes employ lazy initialization of the hw params (initialize it once, get to another path in code, oh no wait! we need to initialize them differently...) (3) Per the spec, calling snd_*_hw_params also makes the stream PREPARED, so calling snd_*_hw_params on a stream that has _either_ been explicitly prepared, _or_ had its hw_params previously set, will trigger this problem. (4) The ALSA<->PulseAudio plugin is an exception to (1) above; i.e., it triggers an assertion when exercising this functionality, as if to say, "I can't do that!" (5) Indeed, the PulseAudio client API forbids re-initialization of the parameters in question on a PulseAudio stream, after instantiation. --The Solution-- Only perform the assertion if snd_pcm_state(io->pcm) != SND_PCM_STATE_PREPARED. The assertion is still useful for cases where the stream is mysteriously non-NULL and the stream has not already been prepared; I can't think of a valid case where: (1) The state is not SND_PCM_STATE_PREPARED, and (2) The underlying PulseAudio stream (pcm->stream) is non-NULL, and (3) snd_pcm_hw_params would be expected to do anything meaningful or correct. For this reason, I kept the assertion in, rather than just removing it (which by itself "fixes" the problem with alsaplayer and my test case, which is submitted with the patch.) --Why It Works-- (1) This assertion might be a throwback from before ioplug was so smart; but apparently, ioplug will now call pulse_prepare, which reinitializes the underlying pulseaudio stream, after pulse_hw_params. (2) There is nothing in pulse_hw_params that actually requires any particular state out of pcm->stream. The underlying pulse stream is simply not used in this call. (3) ioplug catches the fact that the hw_params have been changed, and calls back to pulse_prepare to handle the underlying stream teardown and reinitialization (which pulse_prepare does correctly). --Impact-- (1) At least one open source application, alsaplayer, can now play through the ALSA<->PulseAudio plugin, where it could never do so before. (2) The ALSA<->PulseAudio plugin is now more tightly compatible with the other alsa-lib and alsa-plugin backends (dmix, hw, asym, oss, jack, ...) (3) Older versions of sox might have also suffered from this bug, according to information here: http://www.pulseaudio.org/ticket/120 (although the Ubuntu 7.10 version of sox does not exhibit this behavior). --Notes-- (1) I submitted a first-attempt patch that fixed the problem, but it was inefficient: it needlessly called pulse_prepare _prior_ to the main pulse_hw_params function body. This was a result of my lack of understanding of how ioplug was actually saving me from disaster in the background. Be sure to grab 2_pcm_pulse.c.patch rather than pcm_pulse.c.patch. (2) I tested both positive and negative cases; the ALSA<->PulseAudio plugin works with applications that hit the modified code, as well as applications that don't. I've tested both branches of the conditional I added. Thanks, Sean

2 1

[alsa-devel] snd-hdsp in 2.6.22 crashes badly
by ilya .d 29 Oct '07

29 Oct '07

Hello. i'm using RME Hammerfall HDSP9632.. it worked fine with 2.6.17 and 2.6.20 (-gentoo), i don't think gentoo team modifies alsa tree .. as soon as i upgraded to 2.6.22[-gentoo-r8], it booted fine, but when i started gnome-settigs-daemon, kernell generated OOPS, and i couldn't inload the module with modprobe. i disabled gnome-settings-daemon, and reboot. i managed to use hdspconf , hdspmixer + a couple of applications (puredata and snd editor), it was all good. as soon as i ran alsamixer (well , normaly you don't need it for snd-hdsp, but i wanted to configure my second soundcard, which is SB_live, but forgot to give it '-c 1') - so here the same crash heppends. i tried 2.6.20 back - and all absolutely fine with it. looking at hdsp.c in both kernels i spoted quite a few changes , however i don't know much about it ;[ so i attached the details below. -- cheers, ilya .d

2 2

[alsa-devel] ALSA dma ring buffer issue on 2.6.14
by Filipe Braz 27 Oct '07

27 Oct '07

Hi, I am trying to use ALSA with the OMAP2430 embedded platform, using kernel 2.6.14 for low latency voip applications. The problem seems to be that the driver ring buffer is not filled by the alsa kernel/core on time to be sent to the hw codec. So, If I use the following large buffer size / period-size it works fine: aplay --period-size=1024 --buffer-size=8192 -f S16_LE -c 2 -r 8000 pattern.pcm But if I use smaller buffer / period sizes such as the following it does not work: aplay --period-size=32 --buffer-size=128 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=32 --buffer-size=256 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=64 --buffer-size=512 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=128 --buffer-size=512 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=256 --buffer-size=1024 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=512 --buffer-size=2048 -f S16_LE -c 2 -r 8000 pattern.pcm I am playing a special pattern sound file that I verify before each period is transfered to the codec via dma (on the driver). On smaller buffer sizes. the 1st period AFTER the ringbuffer rotates to the beginning, appears to not contain the pattern data (instead is filled with zeros). This does not happens with larger buffer / periods. On all cases the pattern file is large enough so that the dma ring buffer rotates several times until transfer is complete. I have tested this on the pc and it works just fine. Also if I use the OSS driver for the OMAP2430 instead of the ALSA it works fine. ALSA with OSS emulation works significantly better but there is still some distortion. Any ideas why this happens? Is this a known issue or is there a solution for this problem? Thanks in advance, Filipe

1 0

[alsa-devel] Confused about 'default' device
by Timur Tabi 26 Oct '07

26 Oct '07

I'm having a hard time understand when ALSA does sample rate and format conversion, and when it doesn't. When I issue the following command: mplayer -ao alsa filename.mov -v I get this output: Building audio filter chain for 48000Hz/2ch/s24le -> 0Hz/0ch/s8... [libaf] Adding filter dummy [dummy] Was reinitialized: 48000Hz/2ch/s24le [dummy] Was reinitialized: 48000Hz/2ch/s24le alsa-init: requested format: 48000 Hz, 2 channels, 11 alsa-init: using ALSA 1.0.8 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opend in blocking mode alsa-init: format s24le are not supported by hardware, trying default alsa-lib: pcm_plug.c:729:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' alsa-init: unable to set format: Invalid argument Sure enough, my device can't handle s24le. It can only handle s16be. I was under the impression that ALSA can do format and sample rate conversion if necessary, but I guess that's not always the case. I know ALSA has a bunch of plug-ins for that, but I guess they're not used by default. What do I need to do to tell ALSA to convert s24le to s16be?

4 14

[alsa-devel] New alsa lowlevel driver for the SE-200PCI and SE-90PCI
by sh_okada＠d4.dion.ne.jp 26 Oct '07

26 Oct '07

HI. I mail from Japan. I wrote a driver for the SE-200PCI and SE-90PCI what are salling in Japan. And I want to merge it to current source tree. How should I do? Thanks.

4 16

[alsa-devel] [PATCH] make sbiload to use hwdep interface
by Takashi Iwai 26 Oct '07

26 Oct '07

This patch makes sbiload using the hwdep interface instead of seq instr layer. This simplifies the logic a lot. Takashi diff -r 9b306261208b seq/sbiload/README --- a/seq/sbiload/README Mon Oct 22 15:16:35 2007 +0200 +++ b/seq/sbiload/README Fri Oct 26 16:10:50 2007 +0200 @@ -1,4 +1,4 @@ This is sbiload - an OPL2/3 FM instrumen -This is sbiload - an OPL2/3 FM instrument loader for ALSA sequencer +This is sbiload - an OPL2/3 FM instrument loader for ALSA hwdep Written by Uros Bizjak <uros(a)kss-loka.si> Web: http://www.kss-loka.si/~uros/sbiload.html @@ -14,62 +14,42 @@ By default it will install into /usr/loc By default it will install into /usr/local/bin, change INSTDIR in make.conf to change this. -Before use ----------- - -Please check that the correct sequencer modules are loaded for your -sound card. You need to load snd-synth-opl3 module. - Usage ----- - loadsbi [-p client:port] [-4] [-l] [-P path] [-v level] instfile drumfile + loadsbi [-D device] [-4] [-P path] [-v level] instfile drumfile - -p, --port=client:port - An ALSA client and port number to use + -D, --device=name - An ALSA hwdep name to use -4 --opl3 - Four operators (OPL3) file type - -l, --list - List possible output ports -P, --path=path - Specify the patch path -v, --verbose=level - Verbose level -Because you may have more than one sound card and each sound -card may have several MIDI connections, you have to tell sbiload -which one to use. +When you have multiple sound cards and/or there may be multiple OPL2/3 +synth devices, you have to tell sbiload which one to use, and pass its +hwdep name via -D option. Otherwise, sbiload will load the data onto +the first found device. -First find out what the possibilities are for your system: +To find out the hwdep name of the OPL devices, check +/proc/asound/hwdep file first. It has the list of hwdep devices +currently available on your system. -If you run: - - sbiload -l + % cat /proc/asound/hwdep + 00-00: Emux WaveTable + 00-01: OPL3 FM + 01-00: OPL3 FM -it will give you a list of the devices. On my system I get: - - Port Client name Port name - 64:0 0: MIDI Synth MIDI 0-0 - 65:0 OPL3 FM Synth OPL3 FM Synth port +In the example above, you have two OPL3 devices. To the first one +(00-01), which means the first card (00-) and the second device (01), +pass "-D hw:0,1". For the second OPL3 device (01-00), pass "-D +hw:1,0" instead. - -Only OPLx FM Synth port can be used with sbiload. In the example above -the first one is the external Midi port, and the second is the internal -four-operator FM synthesizer (OPL3). - -If you don't see anything listed then check that the correct ALSA -modules are loaded as in the section "Before use". Running sbiload --------------- -You specify the port to use with the -p option, or by setting the -environment variable ALSA_OUT_PORT. +Pass the instrument and drum file data as arguments. -For example to use port 0 on client 65 to load four-operator FM -instrument - - sbiload -p 65:0 --opl3 std.o3 drums.o3 - -or - - export ALSA_OUT_PORT=65:0 - sbiload --opl3 std.o3 drums.o3 + % sbiload std.o3 drums.o3 Acknowledgements diff -r 9b306261208b seq/sbiload/sbiload.c --- a/seq/sbiload/sbiload.c Mon Oct 22 15:16:35 2007 +0200 +++ b/seq/sbiload/sbiload.c Fri Oct 26 16:10:50 2007 +0200 @@ -1,5 +1,5 @@ /* - * ALSA sequencer SBI FM instrument loader + * ALSA hwdep SBI FM instrument loader * Copyright (c) 2000 Uros Bizjak <uros(a)kss-loka.si> * * This program is free software; you can redistribute it and/or modify @@ -16,11 +16,13 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * + * + * Oct. 2007 - Takashi Iwai <tiwai(a)suse.de> + * Changed to use hwdep instead of obsoleted seq-instr interface */ #include <errno.h> #include <getopt.h> -#include <alsa/sound/ainstr_fm.h> #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -29,24 +31,12 @@ #include <fcntl.h> #include <errno.h> #include <alsa/asoundlib.h> +#include <alsa/sound/asound_fm.h> #include <sys/time.h> #include <unistd.h> -typedef struct sbi_header -{ - char key[4]; - char name[32]; -} -sbi_header_t; - -typedef struct sbi_inst -{ - sbi_header_t header; #define DATA_LEN_2OP 16 #define DATA_LEN_4OP 24 - char data[DATA_LEN_4OP]; -} -sbi_inst_t; /* offsets for SBI params */ #define AM_VIB 0 @@ -59,21 +49,20 @@ sbi_inst_t; #define CONNECTION 10 #define OFFSET_4OP 11 -/* offsets in sbi_header.name for SBI extensions */ -#define ECHO_DELAY 25 -#define ECHO_ATTEN 26 -#define CHORUS_SPREAD 27 -#define TRNSPS 28 -#define FIX_DUR 29 -#define MODES 30 -#define FIX_KEY 31 +/* offsets for SBI extensions */ +#define ECHO_DELAY 0 +#define ECHO_ATTEN 1 +#define CHORUS_SPREAD 2 +#define TRNSPS 3 +#define FIX_DUR 4 +#define MODES 5 +#define FIX_KEY 6 /* Options for the command */ #define HAS_ARG 1 static struct option long_opts[] = { - {"port", HAS_ARG, NULL, 'p'}, + {"device", HAS_ARG, NULL, 'D'}, {"opl3", 0, NULL, '4'}, - {"list", 0, NULL, 'l'}, {"path", HAS_ARG, NULL, 'P'}, {"verbose", HAS_ARG, NULL, 'v'}, {"version", 0, NULL, 'V'}, @@ -83,87 +72,35 @@ static struct option long_opts[] = { /* Number of elements in an array */ #define NELEM(a) ( sizeof(a)/sizeof((a)[0]) ) -#define ADDR_PARTS 4 /* Number of part in a port description addr 1:2:3:4 */ -#define SEP ", \t" /* Separators for port description */ - -#define SBI_FILE_TYPE_2OP 0 -#define SBI_FILE_TYPE_4OP 1 +enum { + FM_PATCH_UNKNOWN, + FM_PATCH_OPL2, + FM_PATCH_OPL3 +}; /* Default file type */ -int file_type = SBI_FILE_TYPE_2OP; +static int file_type = FM_PATCH_OPL2; /* Default verbose level */ -int verbose = 0; +static int verbose = 0; /* Global declarations */ -snd_seq_t *seq_handle; - -int seq_client; -int seq_port; -int seq_dest_client; -int seq_dest_port; +static snd_hwdep_t *handle; #ifndef PATCHDIR #define PATCHDIR "/usr/share/sounds/opl3" #endif -char *patchdir = PATCHDIR; +static char *patchdir = PATCHDIR; /* Function prototypes */ -static void show_list (); -static void show_usage (); -static void show_op (fm_instrument_t * fm_instr); +static void show_usage (void); +static void show_op (struct sbi_patch * instr, int type); -static int load_patch (fm_instrument_t * fm_instr, int bank, int prg, char *name); +static int load_patch (struct sbi_patch * instr); static int load_file (int bank, char *filename); -static int parse_portdesc (char *portdesc); -static int init_client (); - -static void -ignore_errors (const char *file, int line, const char *function, int err, const char *fmt, ...) -{ - /* ignore */ -} - -/* - * Show a list of possible output ports that midi could be sent to. - */ -static void -show_list () { - snd_seq_client_info_t *cinfo; - snd_seq_port_info_t *pinfo; - - int client, err; - - snd_lib_error_set_handler (ignore_errors); - if ((err = snd_seq_open (&seq_handle, "hw", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { - fprintf (stderr, "Could not open sequencer: %s\n", snd_strerror (err)); - return; - } - - printf (" Port %-30.30s %s\n", "Client name", "Port name"); - snd_seq_client_info_alloca(&cinfo); - snd_seq_client_info_set_client(cinfo, -1); - while (snd_seq_query_next_client(seq_handle, cinfo) >= 0) { - client = snd_seq_client_info_get_client(cinfo); - snd_seq_port_info_alloca(&pinfo); - snd_seq_port_info_set_client(pinfo, client); - snd_seq_port_info_set_port(pinfo, -1); - while (snd_seq_query_next_port(seq_handle, pinfo) >= 0) { - unsigned int cap; - - cap = (SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_WRITE); - if ((snd_seq_port_info_get_capability(pinfo) & cap) == cap) { - printf ("%3d:%-3d %-30.30s %s\n", - client, snd_seq_port_info_get_port(pinfo), - snd_seq_client_info_get_name(cinfo), - snd_seq_port_info_get_name(pinfo)); - } - } - } - snd_seq_close (seq_handle); -} +static int init_hwdep (const char *name); /* * Show usage message @@ -172,11 +109,10 @@ show_usage () { show_usage () { char **cpp; static char *msg[] = { - "Usage: sbiload [-p client:port] [-4] [-l] [-P path] [-v level] instfile drumfile", + "Usage: sbiload [-D device] [-4] [-P path] [-v level] instfile drumfile", "", - " -p client:port - A alsa client and port number to send midi to", + " -D device - hwdep device string", " -4 - four operators file type (default = two ops)", - " -l - List possible output ports that could be used", " -P path - Specify the patch path", " -v level - Verbose level (default = 0)", " -V - Show version", @@ -199,169 +135,86 @@ show_version () { * Show instrument FM operators */ static void -show_op (fm_instrument_t * fm_instr) { +show_op (struct sbi_patch * inst, int type) { int i = 0; + int ofs = 0; do { - printf (" OP%i: flags: %s %s %s %s\011OP%i: flags: %s %s %s %s\n", - i, - fm_instr->op[i].am_vib & (1 << 7) ? "AM" : " ", - fm_instr->op[i].am_vib & (1 << 6) ? "VIB" : " ", - fm_instr->op[i].am_vib & (1 << 5) ? "EGT" : " ", - fm_instr->op[i].am_vib & (1 << 4) ? "KSR" : " ", - i + 1, - fm_instr->op[i + 1].am_vib & (1 << 7) ? "AM" : " ", - fm_instr->op[i + 1].am_vib & (1 << 6) ? "VIB" : " ", - fm_instr->op[i + 1].am_vib & (1 << 5) ? "EGT" : " ", - fm_instr->op[i + 1].am_vib & (1 << 4) ? "KSR" : ""); - printf (" OP%i: MULT = 0x%x" "\011\011OP%i: MULT = 0x%x\n", - i, fm_instr->op[i].am_vib & 0x0f, - i + 1, fm_instr->op[i + 1].am_vib & 0x0f); - printf (" OP%i: KSL = 0x%x TL = 0x%.2x\011OP%i: KSL = 0x%x TL = 0x%.2x\n", - i, (fm_instr->op[i].ksl_level >> 6) & 0x03, fm_instr->op[i].ksl_level & 0x3f, - i + 1, (fm_instr->op[i + 1].ksl_level >> 6) & 0x03, fm_instr->op[i + 1].ksl_level & 0x3f); - printf (" OP%i: AR = 0x%x DL = 0x%x\011OP%i: AR = 0x%x DL = 0x%x\n", - i, (fm_instr->op[i].attack_decay >> 4) & 0x0f, fm_instr->op[i].attack_decay & 0x0f, - i + 1, (fm_instr->op[i + 1].attack_decay >> 4) & 0x0f, fm_instr->op[i + 1].attack_decay & 0x0f); - printf (" OP%i: SL = 0x%x RR = 0x%x\011OP%i: SL = 0x%x RR = 0x%x\n", - i, (fm_instr->op[i].sustain_release >> 4) & 0x0f, fm_instr->op[i].sustain_release & 0x0f, - i + 1, (fm_instr->op[i + 1].sustain_release >> 4) & 0x0f, fm_instr->op[i + 1].sustain_release & 0x0f); - printf (" OP%i: WS = 0x%x\011\011OP%i: WS = 0x%x\n", - i, fm_instr->op[i].wave_select & 0x07, - i + 1, fm_instr->op[i + 1].wave_select & 0x07); - printf (" FB = 0x%x, %s\n", - (fm_instr->feedback_connection[i / 2] >> 1) & 0x07, - fm_instr->feedback_connection[i / 2] & (1 << 0) ? "parallel" : "serial"); - i += 2; + unsigned char val; + val = inst->data[AM_VIB + ofs]; + printf (" OP%i: flags: %s %s %s %s", i, + val & (1 << 7) ? "AM" : " ", + val & (1 << 6) ? "VIB" : " ", + val & (1 << 5) ? "EGT" : " ", + val & (1 << 4) ? "KSR" : " "); + val = inst->data[AM_VIB + ofs + 1]; + printf ("\011OP%i: flags: %s %s %s %s\n", i + 1, + val & (1 << 7) ? "AM" : " ", + val & (1 << 6) ? "VIB" : " ", + val & (1 << 5) ? "EGT" : " ", + val & (1 << 4) ? "KSR" : " "); + val = inst->data[AM_VIB + ofs]; + printf (" OP%i: MULT = 0x%x", i, val & 0x0f); + val = inst->data[AM_VIB + ofs + 1]; + printf ("\011\011OP%i: MULT = 0x%x\n", i + 1, val & 0x0f); + + val = inst->data[KSL_LEVEL + ofs]; + printf (" OP%i: KSL = 0x%x TL = 0x%.2x", i, + (val >> 6) & 0x03, val & 0x3f); + val = inst->data[KSL_LEVEL + ofs + 1]; + printf ("\011OP%i: KSL = 0x%x TL = 0x%.2x\n", i + 1, + (val >> 6) & 0x03, val & 0x3f); + val = inst->data[ATTACK_DECAY + ofs]; + printf (" OP%i: AR = 0x%x DL = 0x%x", i, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[ATTACK_DECAY + ofs + 1]; + printf ("\011OP%i: AR = 0x%x DL = 0x%x\n", i + 1, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[SUSTAIN_RELEASE + ofs]; + printf (" OP%i: SL = 0x%x RR = 0x%x", i, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[SUSTAIN_RELEASE + ofs + 1]; + printf ("\011OP%i: SL = 0x%x RR = 0x%x\n", i + 1, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[WAVE_SELECT + ofs]; + printf (" OP%i: WS = 0x%x", i, val & 0x07); + val = inst->data[WAVE_SELECT + ofs + 1]; + printf ("\011\011OP%i: WS = 0x%x\n", i + 1, val & 0x07); + val = inst->data[CONNECTION + ofs]; + printf (" FB = 0x%x, %s\n", (val >> 1) & 0x07, + val & (1 << 0) ? "parallel" : "serial"); + i += 2; + ofs += OFFSET_4OP; } - while (i == (fm_instr->type == FM_PATCH_OPL3) << 1); + while (i == (type == FM_PATCH_OPL3) << 1); printf (" Extended data:\n" " ED = %.3i EA = %.3i CS = %.3i TR = %.3i\n" " FD = %.3i MO = %.3i FK = %.3i\n", - fm_instr->echo_delay, fm_instr->echo_atten, fm_instr->chorus_spread, fm_instr->trnsps, - fm_instr->fix_dur, fm_instr->modes, fm_instr->fix_key); -} - -/* - * Check the result of the previous instr event - */ -static int -check_result (int evtype) { - snd_seq_event_t *pev; - int err; - - for (;;) { - if ((err = snd_seq_event_input (seq_handle, &pev)) < 0) { - fprintf (stderr, "Unable to read event input: %s\n", - snd_strerror (err)); - return -ENXIO; - } - if (pev->type == SND_SEQ_EVENT_RESULT && pev->data.result.event == evtype) - break; - snd_seq_free_event (pev); - } - err = pev->data.result.result; - snd_seq_free_event (pev); - - return err; + inst->extension[ECHO_DELAY], inst->extension[ECHO_ATTEN], + inst->extension[CHORUS_SPREAD], inst->extension[TRNSPS], + inst->extension[FIX_DUR], inst->extension[MODES], + inst->extension[FIX_KEY]); } /* * Send patch to destination port */ static int -load_patch (fm_instrument_t * fm_instr, int bank, int prg, char *name) { - snd_instr_header_t *put; - snd_seq_instr_t id; - snd_seq_event_t ev; +load_patch (struct sbi_patch * inst) { - size_t size; - int err; - - if (verbose > 1) { - printf ("%.3i: [OPL%i] %s\n", prg, fm_instr->type == FM_PATCH_OPL3 ? 3 : 2, name); - show_op (fm_instr); - } - - if ((err = snd_instr_fm_convert_to_stream (fm_instr, name, &put, &size)) < 0) { - fprintf (stderr, "Unable to convert instrument %.3i to stream: %s\n", - prg, snd_strerror (err)); - return -1; - } - memset(&id, 0, sizeof(id)); - id.std = SND_SEQ_INSTR_TYPE2_OPL2_3; - id.prg = prg; - id.bank = bank; - snd_instr_header_set_id(put, &id); - - /* build event */ - memset (&ev, 0, sizeof (ev)); - ev.source.client = seq_client; - ev.source.port = seq_port; - ev.dest.client = seq_dest_client; - ev.dest.port = seq_dest_port; - - ev.flags = SND_SEQ_EVENT_LENGTH_VARUSR; - ev.queue = SND_SEQ_QUEUE_DIRECT; - -__again: - ev.type = SND_SEQ_EVENT_INSTR_PUT; - ev.data.ext.len = size; - ev.data.ext.ptr = put; - - if ((err = snd_seq_event_output (seq_handle, &ev)) < 0) { + ssize_t ret; + ret = snd_hwdep_write(handle, inst, sizeof(*inst)); + if (ret != sizeof(*inst)) { fprintf (stderr, "Unable to write an instrument %.3i put event: %s\n", - prg, snd_strerror (err)); + inst->prog, snd_strerror (ret)); return -1; } - if ((err = snd_seq_drain_output (seq_handle)) < 0) { - fprintf (stderr, "Unable to write instrument %.3i data: %s\n", prg, - snd_strerror (err)); - return -1; - } - - err = check_result (SND_SEQ_EVENT_INSTR_PUT); - if (err >= 0) { - if (verbose) - printf ("Loaded instrument %.3i, bank %.3i: %s\n", prg, bank, name); - return 0; - } else if (err == -EBUSY) { - snd_instr_header_t *remove; - - snd_instr_header_alloca(&remove); - snd_instr_header_set_cmd(remove, SND_SEQ_INSTR_FREE_CMD_SINGLE); - snd_instr_header_set_id(remove, snd_instr_header_get_id(put)); - - /* remove instrument */ - ev.type = SND_SEQ_EVENT_INSTR_FREE; - ev.data.ext.len = snd_instr_header_sizeof(); - ev.data.ext.ptr = remove; - - if ((err = snd_seq_event_output (seq_handle, &ev)) < 0) { - fprintf (stderr, "Unable to write an instrument %.3i free event: %s\n", - prg, snd_strerror (err)); - return -1; - } - - if ((err = snd_seq_drain_output (seq_handle)) < 0) { - fprintf (stderr, "Unable to write instrument %.3i data: %s\n", prg, - snd_strerror (err)); - return -1; - } - - if ((err = check_result (SND_SEQ_EVENT_INSTR_FREE)) < 0) { - fprintf (stderr, "Instrument %.3i, bank %.3i - free error: %s\n", - prg, bank, snd_strerror (err)); - return -1; - } - goto __again; - } - - fprintf (stderr, "Instrument %.3i, bank %.3i - put error: %s\n", - prg, bank, snd_strerror (err)); - return -1; + if (verbose) + printf ("Loaded instrument %.3i, bank %.3i: %s\n", + inst->prog, inst->bank, inst->name); + return 0; } /* @@ -369,66 +222,45 @@ __again: */ static void load_sb (int bank, int fd) { - int len, i; + int len; int prg; - - sbi_inst_t sbi_instr; - fm_instrument_t fm_instr; + struct sbi_patch inst; int fm_instr_type; - len = (file_type == SBI_FILE_TYPE_4OP) ? DATA_LEN_4OP : DATA_LEN_2OP; + len = (file_type == FM_PATCH_OPL2) ? DATA_LEN_4OP : DATA_LEN_2OP; for (prg = 0;; prg++) { - if (read (fd, &sbi_instr.header, sizeof (sbi_header_t)) < (ssize_t)sizeof (sbi_header_t)) + inst.prog = prg; + inst.bank = bank; + + if (read (fd, inst.key, 4) != 4) break; - if (!strncmp (sbi_instr.header.key, "SBI\032", 4) || !strncmp (sbi_instr.header.key, "2OP\032", 4)) { + if (!memcmp (inst.key, "SBI\032", 4) || !memcmp (inst.key, "2OP\032", 4)) { + len = DATA_LEN_2OP; fm_instr_type = FM_PATCH_OPL2; - } else if (!strncmp (sbi_instr.header.key, "4OP\032", 4)) { + } else if (!strncmp (inst.key, "4OP\032", 4)) { + len = DATA_LEN_4OP; fm_instr_type = FM_PATCH_OPL3; } else { - fm_instr_type = 0; if (verbose) printf ("%.3i: wrong instrument key!\n", prg); + fm_instr_type = FM_PATCH_UNKNOWN; } - if (read (fd, &sbi_instr.data, len) < len) + if (read (fd, &inst.name, sizeof(inst.name)) != sizeof(inst.name) || + read (fd, &inst.extension, sizeof(inst.extension)) != sizeof(inst.extension) || + read (fd, &inst.data, len) != len) break; - if (fm_instr_type == 0) + if (fm_instr_type == FM_PATCH_UNKNOWN) continue; - memset (&fm_instr, 0, sizeof (fm_instr)); - fm_instr.type = fm_instr_type; - - for (i = 0; i < 2; i++) { - fm_instr.op[i].am_vib = sbi_instr.data[AM_VIB + i]; - fm_instr.op[i].ksl_level = sbi_instr.data[KSL_LEVEL + i]; - fm_instr.op[i].attack_decay = sbi_instr.data[ATTACK_DECAY + i]; - fm_instr.op[i].sustain_release = sbi_instr.data[SUSTAIN_RELEASE + i]; - fm_instr.op[i].wave_select = sbi_instr.data[WAVE_SELECT + i]; - } - fm_instr.feedback_connection[0] = sbi_instr.data[CONNECTION]; - - if (fm_instr_type == FM_PATCH_OPL3) { - for (i = 0; i < 2; i++) { - fm_instr.op[i + 2].am_vib = sbi_instr.data[OFFSET_4OP + AM_VIB + i]; - fm_instr.op[i + 2].ksl_level = sbi_instr.data[OFFSET_4OP + KSL_LEVEL + i]; - fm_instr.op[i + 2].attack_decay = sbi_instr.data[OFFSET_4OP + ATTACK_DECAY + i]; - fm_instr.op[i + 2].sustain_release = sbi_instr.data[OFFSET_4OP + SUSTAIN_RELEASE + i]; - fm_instr.op[i + 2].wave_select = sbi_instr.data[OFFSET_4OP + WAVE_SELECT + i]; - } - fm_instr.feedback_connection[1] = sbi_instr.data[OFFSET_4OP + CONNECTION]; + if (verbose > 1) { + printf ("%.3i: %s\n", inst.prog, inst.name); + show_op (&inst, fm_instr_type); } - fm_instr.echo_delay = sbi_instr.header.name[ECHO_DELAY]; - fm_instr.echo_atten = sbi_instr.header.name[ECHO_ATTEN]; - fm_instr.chorus_spread = sbi_instr.header.name[CHORUS_SPREAD]; - fm_instr.trnsps = sbi_instr.header.name[TRNSPS]; - fm_instr.fix_dur = sbi_instr.header.name[FIX_DUR]; - fm_instr.modes = sbi_instr.header.name[MODES]; - fm_instr.fix_key = sbi_instr.header.name[FIX_KEY]; - - if (load_patch (&fm_instr, bank, prg, sbi_instr.header.name) < 0) + if (load_patch (&inst) < 0) break; } return; @@ -461,92 +293,57 @@ load_file (int bank, char *filename) { } /* - * Parse port description + * Open a hwdep device */ -static int -parse_portdesc (char *portdesc) { - char *astr; - char *cp; - int a[ADDR_PARTS]; - int count; +static int open_hwdep (const char *name) +{ + int err; + snd_hwdep_info_t *info; - if (portdesc == NULL) - return -1; + if ((err = snd_hwdep_open (&handle, name, SND_HWDEP_OPEN_WRITE)) < 0) + return err; - for (astr = strtok (portdesc, SEP); astr; astr = strtok (NULL, SEP)) { - for (cp = astr, count = 0; cp && *cp; cp++) { - if (count < ADDR_PARTS) - a[count++] = atoi (cp); - cp = strchr (cp, ':'); - if (cp == NULL) - break; - } - - if (count == 2) { - seq_dest_client = a[0]; - seq_dest_port = a[1]; - } else { - printf ("Addresses in %d parts not supported yet\n", count); - return -1; - } + snd_hwdep_info_alloca(&info); + if (!snd_hwdep_info (handle, info)) { + snd_hwdep_iface_t iface = snd_hwdep_info_get_iface(info); + if (iface == SND_HWDEP_IFACE_OPL2 || iface == SND_HWDEP_IFACE_OPL3) + return 0; } - return 0; + snd_hwdep_close(handle); + handle = NULL; + return -EINVAL; } -/* - * Open sequencer, create client port and - * subscribe client to destination port - */ static int -init_client () { - char name[64]; - snd_seq_port_subscribe_t *sub; - snd_seq_addr_t addr; +init_hwdep (const char *name) { + int err; + char tmpname[16]; - if ((err = snd_seq_open (&seq_handle, "hw", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { - fprintf (stderr, "Could not open sequencer: %s\n", snd_strerror (err)); + if (!name || !*name) { + /* auto probe */ + int card = -1; + snd_ctl_t *ctl; + + while (!snd_card_next(&card) && card >= 0) { + int dev; + sprintf(tmpname, "hw:%d", card); + if (snd_ctl_open(&ctl, tmpname, 0) < 0) + continue; + dev = -1; + while (!snd_ctl_hwdep_next_device(ctl, &dev) && dev >= 0) { + sprintf(tmpname, "hw:%d,%d", card, dev); + if (!open_hwdep(tmpname)) + return 0; + } + snd_ctl_close(ctl); + } + fprintf (stderr, "Can't find any OPL3 hwdep device\n"); return -1; } - seq_client = snd_seq_client_id (seq_handle); - if (seq_client < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to determine my client number: %s\n", - snd_strerror (err)); - return -1; - } - - sprintf (name, "sbiload - %i", getpid ()); - if ((err = snd_seq_set_client_name (seq_handle, name)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to set client info: %s\n", - snd_strerror (err)); - return -1; - } - - if ((seq_port = snd_seq_create_simple_port (seq_handle, "Output", - SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_WRITE, - SND_SEQ_PORT_TYPE_SPECIFIC)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to create a client port: %s\n", - snd_strerror (seq_port)); - return -1; - } - - snd_seq_port_subscribe_alloca(&sub); - addr.client = seq_client; - addr.port = seq_port; - snd_seq_port_subscribe_set_sender(sub, &addr); - addr.client = seq_dest_client; - addr.port = seq_dest_port; - snd_seq_port_subscribe_set_dest(sub, &addr); - snd_seq_port_subscribe_set_exclusive(sub, 1); - - if ((err = snd_seq_subscribe_port (seq_handle, sub)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to subscribe destination port: %s\n", - snd_strerror (errno)); + if ((err = open_hwdep (name)) < 0) { + fprintf (stderr, "Could not open hwdep %s: %s\n", name, snd_strerror (err)); return -1; } @@ -558,40 +355,25 @@ init_client () { * and close sequencer */ static void -finish_client () +finish_hwdep () { - snd_seq_port_subscribe_t *sub; - snd_seq_addr_t addr; - int err; - - snd_seq_port_subscribe_alloca(&sub); - addr.client = seq_client; - addr.port = seq_port; - snd_seq_port_subscribe_set_sender(sub, &addr); - addr.client = seq_dest_client; - addr.port = seq_dest_port; - snd_seq_port_subscribe_set_dest(sub, &addr); - if ((err = snd_seq_unsubscribe_port (seq_handle, sub)) < 0) { - fprintf (stderr, "Unable to unsubscribe destination port: %s\n", - snd_strerror (errno)); - } - snd_seq_close (seq_handle); + snd_hwdep_close(handle); + handle = NULL; } /* * Load a .SBI FM instrument patch * sbiload [-p client:port] [-l] [-P path] [-v level] instfile drumfile * - * -p, --port=client:port - An ALSA client and port number to use + * -D, --device=name - An ALSA hwdep name to use * -4 --opl3 - four operators file type - * -l, --list - List possible output ports that could be used * -P, --path=path - Specify the patch path * -v, --verbose=level - Verbose level */ int main (int argc, char **argv) { char opts[NELEM (long_opts) * 2 + 1]; - char *portdesc; + char *name; char *cp; int c; struct option *op; @@ -604,7 +386,7 @@ main (int argc, char **argv) { *cp++ = ':'; } - portdesc = NULL; + name = NULL; /* Deal with the options */ for (;;) { @@ -613,11 +395,11 @@ main (int argc, char **argv) { break; switch (c) { - case 'p': - portdesc = optarg; + case 'D': + name = optarg; break; case '4': - file_type = SBI_FILE_TYPE_4OP; + file_type = FM_PATCH_OPL3; break; case 'v': verbose = atoi (optarg); @@ -625,9 +407,6 @@ main (int argc, char **argv) { case 'V': show_version(); exit (1); - case 'l': - show_list (); - exit (0); case 'P': patchdir = optarg; break; @@ -637,30 +416,14 @@ main (int argc, char **argv) { } } - if (portdesc == NULL) { - portdesc = getenv ("ALSA_OUT_PORT"); - if (portdesc == NULL) { - fprintf (stderr, "No client/port specified.\n" - "You must supply one with the -p option or with the\n" - "environment variable ALSA_OUT_PORT\n"); - exit (1); - } - } - - /* Parse port description to dest_client and dest_port */ - if (parse_portdesc (portdesc) < 0) { - return 1; - } - - /* Initialize client and subscribe to destination port */ - if (init_client () < 0) { + if (init_hwdep (name) < 0) { return 1; } /* Process instrument and drum file */ if (optind < argc) { if (load_file (0, argv[optind++]) < 0) { - finish_client(); + finish_hwdep(); return 1; } } else { @@ -668,15 +431,14 @@ main (int argc, char **argv) { } if (optind < argc) { if (load_file (128, argv[optind]) < 0) { - finish_client(); + finish_hwdep(); return 1; } } else { fprintf(stderr, "Warning: drum file was not specified\n"); } - /* Unsubscribe destination port and close client */ - finish_client(); + finish_hwdep(); return 0; }

1 0

[alsa-devel] [PATCH] Change asound_fm.h in alsa-lib for OPL3 hwdep loader
by Takashi Iwai 26 Oct '07

26 Oct '07

The change for alsa-lib to make sbiload compile. Takashi diff -r f6057aeb5008 include/sound/asound_fm.h --- a/include/sound/asound_fm.h Thu Oct 25 15:36:03 2007 +0200 +++ b/include/sound/asound_fm.h Fri Oct 26 13:09:29 2007 +0200 @@ -29,16 +29,16 @@ #define SNDRV_DM_FM_MODE_OPL2 0x00 #define SNDRV_DM_FM_MODE_OPL3 0x01 -typedef struct snd_dm_fm_info { +struct snd_dm_fm_info { unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ -} snd_dm_fm_info_t; +}; /* * Data structure composing an FM "note" or sound event. */ -typedef struct snd_dm_fm_voice { +struct snd_dm_fm_voice { unsigned char op; /* operator cell (0 or 1) */ unsigned char voice; /* FM voice (0 to 17) */ @@ -60,25 +60,25 @@ typedef struct snd_dm_fm_voice { unsigned char left; /* stereo left */ unsigned char right; /* stereo right */ unsigned char waveform; /* 3 bits: waveform shape */ -} snd_dm_fm_voice_t; +}; /* * This describes an FM note by its voice, octave, frequency number (10bit) * and key on/off. */ -typedef struct snd_dm_fm_note { +struct snd_dm_fm_note { unsigned char voice; /* 0-17 voice channel */ unsigned char octave; /* 3 bits: what octave to play */ unsigned int fnum; /* 10 bits: frequency number */ unsigned char key_on; /* set for active, clear for silent */ -} snd_dm_fm_note_t; +}; /* * FM parameters that apply globally to all voices, and thus are not "notes" */ -typedef struct snd_dm_fm_params { +struct snd_dm_fm_params { unsigned char am_depth; /* amplitude modulation depth (1=hi) */ unsigned char vib_depth; /* vibrato depth (1=hi) */ unsigned char kbd_split; /* keyboard split */ @@ -90,17 +90,17 @@ typedef struct snd_dm_fm_params { unsigned char tomtom; unsigned char cymbal; unsigned char hihat; -} snd_dm_fm_params_t; +}; /* * FM mode ioctl settings */ -#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, snd_dm_fm_info_t) +#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, struct snd_dm_fm_info) #define SNDRV_DM_FM_IOCTL_RESET _IO ('H', 0x21) -#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, snd_dm_fm_note_t) -#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, snd_dm_fm_voice_t) -#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, snd_dm_fm_params_t) +#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, struct snd_dm_fm_note) +#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, struct snd_dm_fm_voice) +#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, struct snd_dm_fm_params) #define SNDRV_DM_FM_IOCTL_SET_MODE _IOW('H', 0x25, int) /* for OPL3 only */ #define SNDRV_DM_FM_IOCTL_SET_CONNECTION _IOW('H', 0x26, int) @@ -112,4 +112,21 @@ typedef struct snd_dm_fm_params { #define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24 #define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25 +/* + * Patch Record - fixed size for write + */ + +#define FM_KEY_SBI "SBI\032" +#define FM_KEY_2OP "2OP\032" +#define FM_KEY_4OP "4OP\032" + +struct sbi_patch { + unsigned char prog; + unsigned char bank; + char key[4]; + char name[25]; + char extension[7]; + unsigned char data[32]; +}; + #endif /* __SOUND_ASOUND_FM_H */

1 0

[alsa-devel] HDA: Add support for Samsung Q1 Ultra Vista edition.
by Tobin Davis 26 Oct '07

26 Oct '07

This patch adds full record and playback support for the Samsung Q1 Ultra - Vista model (different codec than the earlier Q1 Ultra models). Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] [PATCH] AD1986A 6-channel bugs
by zhejiang 26 Oct '07

26 Oct '07

Hi, I have a ICH8M + AD1986A machine. In 6 channel modes, if adjust the volumes when playing 2-channel or 4-channel music, codec will be muted and all widget's Amp-Out vals will be reset. If you play 2-channel files in 6 channel modes,there is noise in the jacks. By doing experiments, I found that the 6-channel bug was caused by sharing channels. If play 2-channel audio file,the rest channels will share channels with the first 2-channel. The HDA spec doesn’t define the exact behavior under such circumstances. Majority of the codec chips support sharing channel, but seem that AD1986A doesn’t support. I wrote a patch and tested it in my hardware,now it works well. diff -Nur a/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c b/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c --- a/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c 2007-10-25 10:03:29.000000000 -0400 +++ b/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c 2007-10-25 10:02:46.000000000 -0400 @@ -2506,7 +2506,8 @@ if (chs >= (i + 1) * 2) /* independent out */ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2, format); - else /* copy front */ + else if (codec->vendor_id != 0x11d41986) + /* copy front if it is NOT AD1986A*/ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } In alsa bugzilla and maillist, I found a lot of issues about AD1986A 6-channel .Anybody know the detail about AD1986A codec chip and give any comments? Here is a old description of the issue. > > Hi, > > I have a ICH8M + AD1986A machine and run alsa-driver-1.0.15rc. > > I found that : > In 6 channel mode,if adjust the front or pcm volume when playing music, > all channel will be muted,but if adjust the volume alone(don't play > music at the same time),every thing is ok. > I use printk to display the verb in the azx_send_cmd(), > when adjusting the volume in either case, there are just two verb > (0x300) sent to the nid(0x1b) [correct action]. > But in the first case, the dac widget amp-out vals are all affected. > > > Some other interesting thing, > If i adjust the Sur or CLFE volume when playing music, the printk show > that correct verb is sent, but the codec status don't change. > If I adjust them alone,every thing is ok.

2 3

[alsa-devel] aplay appication failed to open the alsa device
by Tharmarajan G 24 Oct '07

24 Oct '07

Hi, I am working on a Development Board which has the PXA270 processor and TSC2301 Audio Codec Chip. The TSC2301 Chip is interfaced with the processor by SPI interface for reading/writing the TSC2301's registers. The TSC2301 Chip is interfaced with the processor by I2S interface for sending/receiving Audio Data to/from the TSC2301 Audio Codec Chip. I have downloaded the pxa-uda1380.c alsa driver source for uda1380 audio codec chip and modified it for tsc2301 . I am using the kernel version 2.6.22 and cross-compiler version arm-linux-gcc 3.4.1. I have built the kernel image with ALSA audio support and the applications aplay and arecord for playing and recording audio data using ALSA api. I have got the alsa related entries in the /proc filesystem . But when playing the aplay, it failed to open the device. [root@SIRIUS /flash2]#./aplay /usr/local/Songs/AjRaj-Anthimanthirai-Theme4.mp3 ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:545: audio open error: No such file or directory [root@SIRIUS /flash2]# /aplay -D /dev/snd/pcmC0D0p /usr/local/Songs/AjRaj-Anthimanthirai-Theme4.mp3 ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM /dev/snd/pcmC0D0p aplay: main:545: audio open error: No such file or directory What could be the issue ? Am I missing anyting in the driver for alsa ? Regards, Tharmarajan G

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