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[alsa-devel] cmipci CM_FLINKOFF disables multinchannel playback
by Timofei V. Bondarenko 30 Oct '07

30 Oct '07

Hi. I found the latest cmipci driver cannot playback in multichannel mode anymore. My hardware is 'CMI8738 model 55' attempting to play 2 or 4 channels through the 2nd DAC i get no sound at all. 6 ch produces sound only at center/bass while front and rear still mute. I found the root of problem in redefinition of some bits of the CM_REG_MISC_CTRL (0x18) register. Was: CM_FLINKON=0x40, CM_FLINKOFF=0x20 Now it is: CM_FLINKON=0x80 CM_FLINKOFF=0x40 Note the CM_FLINKOFF does collide with the CM_MUTECH1=0x40 bit. This bit disables playback on 2nd DAC. The CM_FLINKOFF/CM_FLINKON bits used in the 'Modem' mixer control only. During startup the mixer turns the modem off somewhere and this disables 2nd DAC as well. Complete removal of this 'Modem' control fixes the problem. Is it an acceptable solution? I guess the snd_cmipci_control_switches could be removed too. Regards. Tim. --- alsa-driver-hg20071027/alsa-kernel/pci/cmipci.c 2007-09-18 04:00:10.000000000 +0400 +++ alsa-driver/alsa-kernel/pci/cmipci.c 2007-10-29 17:09:31.000000000 +0300 @@ -2389,7 +2389,6 @@ DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MIS // DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0); // DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0); // DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */ -DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0); #define DEFINE_SWITCH(sname, stype, sarg) \ { .name = sname, \ @@ -2597,7 +2596,7 @@ static struct snd_kcontrol_new snd_cmipc /* card control switches */ static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ - DEFINE_CARD_SWITCH("Modem", modem), +// DEFINE_CARD_SWITCH("Modem", modem), };

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[alsa-devel] PATCH: ln2440sbc ac97 support
by Ivan Kuten 29 Oct '07

29 Oct '07

Hello, This patch adds ac97 support for ln2440sbc board from LittleChips. This board is based on s3c2440 SoC + AC97 Realtek ALC650 codec. Existing s3c2443 implementation is slightly modified because s3c2440 and s3c2443 have different AC97 interrupts. Regards, Ivan Signed-off-by: Ivan Kuten <ivan.kuten(a)promwad.com> diff -urN linux-2.6.23/sound/soc/s3c24xx/Kconfig linux-2.6.23-2440/sound/soc/s3c24xx/Kconfig --- linux-2.6.23/sound/soc/s3c24xx/Kconfig 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/Kconfig 2007-10-28 01:18:23.000000000 +0200 @@ -34,4 +34,12 @@ Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC + select SND_S3C2443_SOC_AC97 + select SND_SOC_AC97_CODEC + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. diff -urN linux-2.6.23/sound/soc/s3c24xx/Makefile linux-2.6.23-2440/sound/soc/s3c24xx/Makefile --- linux-2.6.23/sound/soc/s3c24xx/Makefile 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/Makefile 2007-10-25 01:28:18.000000000 +0200 @@ -10,6 +10,8 @@ # S3C24XX Machine Support snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff -urN linux-2.6.23/sound/soc/s3c24xx/ln2440sbc_alc650.c linux-2.6.23-2440/sound/soc/s3c24xx/ln2440sbc_alc650.c --- linux-2.6.23/sound/soc/s3c24xx/ln2440sbc_alc650.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.23-2440/sound/soc/s3c24xx/ln2440sbc_alc650.c 2007-10-28 00:27:41.000000000 +0200 @@ -0,0 +1,86 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten(a)promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-ac97.h" + +static struct snd_soc_machine ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c2443_ac97_dai[0], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_machine ln2440sbc = { + .name = "LN2440SBC", + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .machine = &ln2440sbc, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff -urN linux-2.6.23/sound/soc/s3c24xx/s3c2443-ac97.c linux-2.6.23-2440/sound/soc/s3c24xx/s3c2443-ac97.c --- linux-2.6.23/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-28 00:55:57.000000000 +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> @@ -253,7 +253,7 @@ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq, + ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, IRQF_DISABLED, "AC97", NULL); if (ret < 0) { printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); @@ -266,7 +266,7 @@ static void s3c2443_ac97_remove(struct platform_device *pdev) { - free_irq(IRQ_S3C2443_AC97, NULL); + free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); clk_put(s3c24xx_ac97.ac97_clk); iounmap(s3c24xx_ac97.regs); diff -urN linux-2.6.23/sound/soc/s3c24xx/s3c24xx-ac97.h linux-2.6.23-2440/sound/soc/s3c24xx/s3c24xx-ac97.h --- linux-2.6.23/sound/soc/s3c24xx/s3c24xx-ac97.h 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/s3c24xx-ac97.h 2007-10-28 00:32:04.000000000 +0200 @@ -20,6 +20,12 @@ #define AC_CMD_ADDR(x) (x << 16) #define AC_CMD_DATA(x) (x & 0xffff) +#ifdef CONFIG_CPU_S3C2440 +#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97 +#else +#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 +#endif + extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/

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[alsa-devel] [PATCH] ac97_patch: compilation warning fix
by Krzysztof Helt 29 Oct '07

29 Oct '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch kills these two compilation warnings if power management is disabled: sound/pci/ac97/ac97_patch.h:86: warning: 'snd_ac97_restore_status' declared 'static' but never defined sound/pci/ac97/ac97_patch.h:87: warning: 'snd_ac97_restore_iec958' declared 'static' but never defined Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The patch was done against kernel 2.6.24-rc1-git4. diff -urp linux-ref/sound/pci/ac97/ac97_patch.h linux-new/sound/pci/ac97/ac97_patch.h --- linux-ref/sound/pci/ac97/ac97_patch.h 2007-10-28 11:11:20.000000000 +0100 +++ linux-new/sound/pci/ac97/ac97_patch.h 2007-10-28 12:46:34.000000000 +0100 @@ -83,8 +83,10 @@ static int snd_ac97_swap_ctl(struct snd_ const char *s2, const char *suffix); static void snd_ac97_rename_vol_ctl(struct snd_ac97 *ac97, const char *src, const char *dst); +#ifdef CONFIG_PM static void snd_ac97_restore_status(struct snd_ac97 *ac97); static void snd_ac97_restore_iec958(struct snd_ac97 *ac97); +#endif static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol,

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[alsa-devel] [PATCH] Dreamcast AICA sound - Get rid of annoying compiler warning
by Adrian McMenamin 29 Oct '07

29 Oct '07

This patch supresses an annoying compiler warning that the variable err may be used uninitialised. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 88dc840..8861d2f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -237,6 +237,7 @@ static int aica_dma_transfer(int channels, int buffer_size, struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; unsigned long flags; + err = 0; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels);

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[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

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[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

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[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?

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[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.

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[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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