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

[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; }

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[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 */

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[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>

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[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.

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[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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[alsa-devel] New hardware report
by Alain DANGER 24 Oct '07

24 Oct '07

Hi. I managed to get the sound work with ALSA on the following MB : Shuttle AK32VN. OS : Kubuntu 7.10. Regards. Alain Danger

2 1

[alsa-devel] mplayer plays video twice as fast with Alsa than with OSS
by Timur Tabi 24 Oct '07

24 Oct '07

Hi, I wrote an Alsa device driver that supposedly works with the OSS emulation, but one of our customers is reporting that mplayer with the "-ao oss" option plays video twice at double speed. If they select "-ao alsa" instead, then it plays normally. Is this a known problem?

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Re: [alsa-devel] [Jackit-devel] RME HDSPe floating point question
by Didier Villevalois 24 Oct '07

24 Oct '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, (Cross-posting my report - last time i do this) Great news (almost for me in anyways). RME HDSPe PCI *works as is* with the current driver. Firmware uploads as well. I had nothing to do to make it work! (I think the ExpressCard should work as well...) Maybe the alsa-project wiki page can be updated. Little by little in the next weeks, I will delve into the driver's code to identify parts that needs to be enhanced in order to take profit of the 32bit floating point format that the HDSPe card can provide. When ready, I'll contact RME to request them for informations about the activation of this mode. Didier. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHcG6VE1mALQL3JsRAjeZAKCJqicClHgy1UDc1YvAywuw1FokyQCeJO2d BhWpkxL8xLGxKB3zmbWFQpk= =wpbm -----END PGP SIGNATURE-----

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[alsa-devel] [RFC] ALSA scenario API
by Liam Girdwood 23 Oct '07

23 Oct '07

A little late with this but "better late than never".....I did promise a few people that I would post what was imho needed for ALSA scenarios. Also, apologies for cross posting, but it some of this probably applies to both lists... ALSA probably needs functionality to define and set audio usage scenarios. A scenario is simply a list of sound card mixer settings for a given use case. This is quite important for mobile devices (e.g. Smartphones, GPS, PDA, etc) with complex audio routing and power requirements. Some example scenarios could be :- "HiFi Headphone Playback" "HiFi Speaker Playback" "Handset Phone Call" "Headset Phone Call" etc... Scenarios are changed atm by either manually setting each control (e.g. using alsamixer or alsa-lib in apps) or by using alsactl to restore a scenario. The above two methods are not very portable between devices and applications. We would ideally like to just say "set Phone Call Handset" for a regular handset phone call or "set HiFi Headphones" for MP3 playback. ALSA also needs to have functionality to alias mixer controls. This would allow a single master volume/mute control to be exposed to user space and have it aliased to different hardware controls based upon the current scenario. e.g. Many mobile devices have different physical volume controls for speaker and headphones. This makes it somewhat difficult for user space apps to change the correct volume (without being bound to the hardware). We would like to be able to just say "set volume to 0dB" and have the alias take care of the correct physical CODEC mixer for the current scenario. Proposal: ======== 1. Add some new scenario functions to (s)alsa-lib. e.g. snd_scenario_set(); snd_scenario_get(); snd_scenario_list(); _set() would set the scenario e.g. snd_scenario_set(card, SND_VOICE_CALL_HANDSET); We would need to have a list of standard scenarios so that applications could be more portable. Vendor scenario's should also be supported. _get() would return the current scenario. _list() would return a list of supported scenarios for the given device. 2. Add some new mixer alias functions to set/get the current "master" volume/mute. e.g. snd_kcontrol *snd_scenario_volume_set(card, type, volume); snd_kcontrol *snd_scenario_switch_set(card, type, switch); where type is playback, capture, sidetone, bypass, etc This would set the volume for the current scenario and would guarantee always to get/set the _correct_ volume/switch. 3. Add a new function to return the active PCM for a given scenario. e.g. snd_scenario_get_pcm(card, scenario); Would return the given PCM for a particular scenario. i.e. the WM9713 driver has PCM 0.0 for high quality (higher power) audio (MP3), PCM 0.1 for lower power voice and PCM 0.2 for lowest power and quality system sounds. Note: scenarios may need to be more tightly bound to PCM's rather than cards. 4. Define a file format to represent each scenario. Probably base this upon alsactl's format as alsamixer could be used to quickly generate good settings and then save them with alsactl. Alias, scenario names and PCM would need to be added later somehow. 5. (Optional) Allow scenarios to define active kcontrols. This would allow tools like alsamixer to only display mixer controls for the given scenario. Controls that don't effect the scenario would not be rendered. (similar to alsamixers Playback|Capture|All tab) 6. (OHM list) Add input driver (or similar) for sound card user events. We need to be able to send events like "Headphone Jack Insertion", "Mic Jack Insertion", etc to user space. This would allow changing of scenario and power configuration depending on policy and configuration. Just my 2p Unfortunately I'm not in a position to implement any of this functionality. However, it's probably useful to folks working on mobile devices. Maybe a good project for Google SoC. HTH Liam

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Re: [alsa-devel] [PATCH] asihpi checkpatch clean plus control name refactor
by Takashi Iwai 23 Oct '07

23 Oct '07

At Tue, 23 Oct 2007 16:48:20 +1300, Eliot Blennerhassett wrote: > > checkpatch.pl --no-tree now reports no errors on asihpi.c > Redo control names to be strings only (not using control index) > > Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> Thanks, applied to HG tree now. (BTW, I couldn't access the ftp. Anon-ftp isn't allowed?) Takashi

1 0

[alsa-devel] [PATCH] asihpi checkpatch clean plus control name refactor
by Eliot Blennerhassett 23 Oct '07

23 Oct '07

checkpatch.pl --no-tree now reports no errors on asihpi.c Redo control names to be strings only (not using control index) Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> Patch here; (70k) ftp://ftp.audioscience.com/internet/download/beta/asihpi_checkpatch_clean.d… regards Eliot

1 0

[alsa-devel] USB device gives wrong data?
by william estrada 23 Oct '07

23 Oct '07

Hi group, I think I have found a problem with the USB sound driver? It appears that the current drive returns a x'81' for silence but it should be a x'7f'. When I use my laptop's sound device and record without a mic, I get a file full of x'7f's. If I do the same with my cheap-o USB device I get a file full of x'81's. The sound quality from the laptop is unacceptable, so that is way I want to use the USB device. I have written a recording program that I have a 'silence' filter that filters out segments of dead air. It works with the laptop's sound device but not the USB device. The filter works by remove x'7f's after allowing a selectable amount. I get the results using arecord which does not have the filtering. The USB device is a C -Media Electronics, Inc. I am running FC6. > [root@laptop ~]# lsmod |grep snd > snd_usb_audio 81985 0 > snd_usb_lib 21057 1 snd_usb_audio > snd_rawmidi 28481 1 snd_usb_lib > snd_hwdep 13637 1 snd_usb_audio > snd_ali5451 27853 1 > snd_ac97_codec 99813 1 snd_ali5451 > snd_ac97_bus 6593 1 snd_ac97_codec > snd_seq_dummy 8133 0 > snd_seq_oss 37057 0 > snd_seq_midi_event 11841 1 snd_seq_oss > snd_seq 57009 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event > snd_seq_device 12365 4 snd_rawmidi,snd_seq_dummy,snd_seq_oss,snd_seq > snd_pcm_oss 46433 0 > snd_mixer_oss 20545 1 snd_pcm_oss > snd_pcm 80965 4 snd_usb_audio,snd_ali5451,snd_ac97_codec,snd_pcm_oss > snd_timer 26693 2 snd_seq,snd_pcm > snd 58181 140 snd_usb_audio,snd_rawmidi,snd_hwdep,snd_ali5451,snd_ac97_codec,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > soundcore 12193 1 snd > snd_page_alloc 14281 1 snd_pcm Where do I report this 'bug' or how do I fix it? Thanks for your time. Sorry for the multiple posting, not sure where is should have gone. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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Re: [alsa-devel] WM9712 monoout
by Mike Rapoport 22 Oct '07

22 Oct '07

Liam Girdwood wrote: > On Sun, 2007-10-21 at 09:35 +0200, Mike Rapoport wrote: >> Hello, >> >> I'm trying to enable phone on a board with WM9715 codec. >> I have not found a control responsible for mute/unmute the MONOUT. >> My question is whether I should add this control as platform specific control, >> or it would be correct to patch sound/soc/codecs/wm9712.c? > > It should probably go in the codec driver. It looks like an omission. > >> Also, "Mono Playback Volume" control is not inverted, which seems to be a bug. >> > > Could you send a patch ? Sure. A also found that treble and bass need to be inverted. > Many Thanks > > Liam > Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> sound/soc/codecs/wm9712.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 986b5d5..427cb61 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -102,7 +102,8 @@ SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), @@ -145,8 +146,8 @@ SOC_ENUM("Bass Control", wm9712_enum[5]), SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), -SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), +SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), +SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),

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[alsa-devel] snd-pcsp doesn't accept the index parameter
by Alexander E. Patrakov 22 Oct '07

22 Oct '07

It is impossible to put pcsp at a stable index without the trivial patch below. Signed-Off-By: Alexander E. Patrakov <patrakov(a)ums.usu.ru> --- pcsp.c +++ pcsp.c @@ -27,6 +27,8 @@ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); -- Alexander E. Patrakov

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[alsa-devel] [PATCH] hda: STAC9228 updated DMUX nid.
by Matthew Ranostay 22 Oct '07

22 Oct '07

Changed the dmux for STAC9228 from ADC1MUX to ADC0MUX to avoid confusion. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 7cf5e23f804e pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Oct 19 08:23:00 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Oct 19 13:47:35 2007 -0400 @@ -2777,7 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1c; + spec->dmux_nid = 0x1b; break; default: spec->num_dmics = 0;

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[alsa-devel] SALSA Lib - TLV Deprecated warnings
by J. Scott Merritt 22 Oct '07

22 Oct '07

It appears that if SALSA Lib 0.0.12 is built -without- TLV enabled in the configuration options, then simply -including- asoundlib.h from an application program will trigger deprecated warnings on some of the TLV read/write function definitions (even though they are not called from the application). Is there something we can do to the header files to eliminate these warnings ? Thanks, Scott.

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[alsa-devel] alsa-lib not installing plugins
by Larry D. Pyeatt 22 Oct '07

22 Oct '07

Hi, I wrote a new plugin for alsa-lib. After running make, there are .o and .lo files in src/pcm. However, when I run make install, nothing gets put into /usr/lib/alsa-lib. I first tried installing alsa-lib-1.0.15rc3. Then I tried alsa-lib-1.0.13rc3 and alsa-lib-1.0.14rc4. All of them did the same thing. Am I missing something obvious? Since they are all doing the same thing, it must be either that I just don't know what I'm doing, or something else in my system is not working. I would appreciate any suggestions on what could cause this behavior. Thanks, -- Larry D. Pyeatt, PhD. pyeatt(a)cs.ttu.edu http://www.cs.ttu.edu/~pyeatt

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 21 Oct '07

21 Oct '07

Hi, I understand why this happens. Your sound card has a single "headphone" output, and since it is single, it is called "Master" the stac you have also has a real master volume control, that controls all the DACs, not just the headphones. I added support for it, thus two controls collided. The following patch will rename the "Master" output to "Headphone" Note that on the stac 9205/9204, the only one that has no master volume (the volume knob) but has a "Master Volume" amplifier widget, this will probably rename the "Master Volume" to Headphones. This chip is quite different, thus I think this should be handled in its patch function. I see what I can do later. Meanwhile the rename from "Master" to Headphones" is the only regression. Best regards, Maxim Levitsky PS: I reply to this mail with the patch.

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Jan-Simon Möller 21 Oct '07

21 Oct '07

I've also no sound. Had it working using the alsa.git tree _before_ the last merge (>48h). Needed to use model=asus for this fujitsu v5505 here. Now it doesn't work, even with other model-options (fujitsu, laptop). legolas:/proc/asound # cat cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc400000 irq 22 legolas:/proc/asound # cat pcm 00-01: Conexant Digital : Conexant Digital : playback 1 00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1 legolas:/proc/asound # cat hwdep 00-00: HDA Codec 0 legolas:/proc/asound # cat devices 2: : timer 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: : sequencer legolas:/proc/asound # cat card0/codec#0 Codec: Conexant CX20549 (Venice) Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x1734110e Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x140]: 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19 0x17* Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19 0x17* Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19 0x17* Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x40]: 48000 bits [0x6]: 16 20 formats [0x5]: PCM AC3 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates [0x540]: 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14* 0x12 0x11 0x15 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Best regards Jan-Simon

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Re: [alsa-devel] [Jackit-devel] RME HDSP driver enhancements for HDSPe cards
by John Rigg 19 Oct '07

19 Oct '07

On Fri, Oct 19, 2007 at 08:58:00AM +0100, Ed W wrote: > RME are also helpful to the extent that you ask them specific questions > and in the name of writing drivers. They should give you hardware > details, etc > > > As an aside, apart from CPU load I will be suprised if turning floats > into 32bit ints makes any difference to the audio, but I agree that it > would be a nice feature to have! IIRC the new RME cards convert 24 bit integers to 32 bit floats in hardware (and vice versa), thus saving apps like jackd (and the CPU) the trouble. jackd doesn't support that yet, but AFAIK it shouldn't be too much effort to change that. John

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[alsa-devel] [PATCH] hda: Add dmux to STAC 9228
by Matthew Ranostay 19 Oct '07

19 Oct '07

Added a dmux to the STAC9228 cards with DMIC support. And added a STAC_DIGITAL_INPUT_SOURCE macro for repeating mixer code. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 821a4558dd47 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Oct 18 17:38:17 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Oct 18 17:35:56 2007 -0400 @@ -430,6 +430,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_DIGITAL_INPUT_SOURCE(cnt) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Digital Input Source", \ + .count = cnt, \ + .info = stac92xx_dmux_enum_info, \ + .get = stac92xx_dmux_enum_get, \ + .put = stac92xx_dmux_enum_put,\ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -482,14 +492,7 @@ static struct snd_kcontrol_new stac925x_ }; static struct snd_kcontrol_new stac9205_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - .count = 1, - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, - }, + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), STAC_VOLKNOB(0x24), @@ -521,6 +524,7 @@ static struct snd_kcontrol_new stac922x_ static struct snd_kcontrol_new stac927x_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), @@ -2773,6 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; + spec->dmux_nid = 0x1c; break; default: spec->num_dmics = 0;

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[alsa-devel] ALSA Bluetooth connection issue
by 19 Oct '07

19 Oct '07

Hi All: Im using ALSA 1.0.14 and bluez 3.20 and linux 2.6.18.The linux 2.6.18 (montavista) is running on an embeded development board(using S3C2410). When i tried to use 'aplay' to connect my bluetooth headset, i got the following error from ALSA: $ aplay -D bluetooth /1.wav audio[326]: Accepted new client connection on unix socket (fd=8) audio[326]: Package PKT_TYPE_CFG_REQ:0 hcid[318]: /org/bluez: org.bluez.Manager.FindAdapter() audio[326]: Got path /org/bluez/hci0 for adapter with address 00:80:E1:00:00:00 hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: no matching session found. creating a new one hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache audio[326]: Audio service discovery completed with success hcid[318]: /org/bluez/hci0: org.bluez.Adapter.FinishRemoteServiceTransaction() hcid[318]: found matching session, removing from list audio[326]: Found Hansfree record audio[326]: Discovered Headset service on RFCOMM channel 1 audio[326]: Found Headset record audio[326]: Found Audio Sink audio[326]: Selecting default device audio[326]: avdtp_ref(0x3b018): ref=2 audio[326]: a2dp_source_request_stream: selected SEP 0x37250 audio[326]: avdtp_ref(0x3b018): ref=3 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[326]: session_cb audio[326]: DISCOVER request succeeded audio[326]: seid 1 type 1 media 0 in use 0 audio[326]: seid 2 type 1 media 0 in use 0 audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 1 type 1 media 0 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 2 type 1 media 0 audio[326]: Discovery complete audio[326]: avdtp_set_configuration(0x3b018): int_seid=1, acp_seid=1 audio[326]: session_cb audio[326]: SET_CONFIGURATION request succeeded audio[326]: SBC Source: Set_Configuration_Cfm audio[326]: avdtp_ref(0x3b018): ref=4 audio[326]: stream state changed: IDLE -> CONFIGURED audio[326]: session_cb audio[326]: OPEN request succeeded audio[326]: stream state changed: CONFIGURED -> OPEN audio[326]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[326]: SBC Source: Open_Cfm audio[326]: session_cb audio[326]: START request succeeded audio[326]: SBC Source: Start_CAlignment trap: bluetoothd-serv (326) PC=0x0001be44 Instr=0xe1c320b0 Address=0xbec85c2d FSR 0x813 fm audio[326]: SBC Source SEP 0x37250 locked hcid[318]: Service owner exited: :1.2 aplay: main:550: audio open error: Input/output error hcid[318]: Audio service (audio) was killed by signal 7 ALSA's aplay reports an "Input/output error". Later i digged into alsa-lib-1.0.14/src/pcm/pcm.c and found that snd_config_search_definition() returns an error. It seems ALSA didn't find the proper conf definition or something like that. My ".asoundrc" is as followed: pcm.a2dpd { type a2dpd } pcm.bluetooth { type bluetooth device 00:0D:3C:EB:53:F0 } pcm.!default { type hw card 0 device 0 } pcm.card0 { type hw card 0 } ctl.card0 { type hw card 0 } Does someone have any idea about what's going wrong in this? Thanks in advance! semiyd _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[alsa-devel] ALSA-for custom Operating System????
by Anoop Thomas 19 Oct '07

19 Oct '07

First of all don't make fun of me if i am asking a foolishness.Could you plz tell me can i use alsa for my custom OS which concentrates more on audio side.It doesn't uses linux kernel.Does it make a difference???? (ie. is it enough to give sufficient data to ALSA to help me to make the audio out of the system).If I can plz give me a brief description.... Help me plz................. --------------------------------- Forgot the famous last words? Access your message archive online. Click here.

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[alsa-devel] [PATCH] s3c2443-ac97: compilation fix
by Krzysztof Helt 18 Oct '07

18 Oct '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The Samsung S3C24xx uses new architecture file layout in the post 2.6.23 kernel. This patch fixes include path for the s3c2443-ac97.c. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The patch was done against the 2.6.23-git13 kernel. diff -urp linux-ref/sound/soc/s3c24xx/s3c2443-ac97.c linux-new/sound/soc/s3c24xx/s3c2443-ac97.c --- linux-ref/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-new/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-18 22:27:43.547470033 +0200 @@ -32,7 +32,7 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-ac97.h> +#include <asm/plat-s3c/regs-ac97.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> ---------------------------------------------------------------------- Jaros�aw Drob, Przyjdziemy i Spie***ymy - poznaj kandydata! Kliknij >>> http://link.interia.pl/f1bef

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[alsa-devel] Kernel hierarchy
by Adrian McMenamin 18 Oct '07

18 Oct '07

Why are sound drivers in /sound and not /drivers/sound? Ok, not the world's most pressing problem, but enquiring minds want to know! Adrian

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